## How the JACK Sound Daemon is capable of running at 192kHz

Most of my Linux-computers have as their sound-server ““. But specifically on my laptop named ‘‘, I have set up the to be able to run as an alternative, yet not to be running by default. I have performed experiments on that laptop, to confirm that I can launch this sound-server, using a GUI named ‘‘, but have also had to make modifications to how this GUI executes commands from the user, so that its start-up pauses the , which has been able to resume successfully after I was done using . Without such a detail, the attempt should not be made.

One fact which I can see in , is that is capable of running at 192kHz, even though it has not interrogated any of the available devices, about their real capabilities are.

The reason this is possible is the fact that individual sound devices are just clients to that daemon, including any number of devices that act as sound-sources, rather than acting as sinks, i.e. that act as inputs rather than as one output.

I also own a USB-Sound-Device named the ‘‘, which is mainly intended for use in sound capture, but which also has outputs intended for monitoring purposes.

If I was to run at 192kHz, then one simple consequence of that would be, that zero actual sound-devices would remain compatible with it. As to how cleanly an attempt to connect to an incompatible device exits, giving error messages or crashes, I have not tested, because when I tested the , I took into account the real limit of that device at 96kHz.

Similarly, the runs with 32-bit linear precision by default. In this case, when we enable devices to act as clients, which are only capable of 24-bit sample-depth, which is common, the mismatch is safely ignored. already sees to it, that the last 8 bits of precision get ignored.

Now, I could be cautious and worry, that because of errors in the Linux drivers, those last 8 bits somehow get mapped to a control register as an error. But then the simple way to test for that, was simply to send some 32-bit sound through JACK, to this output device. What I found when testing this, is that the basic operation of the was not disturbed, even though my hearing was not good enough, to tell me when I had my Sennheisers on, whether in fact 24-bit precision was still working. I was mainly testing, that trying to send a 32-bit value, does not disrupt the actual operation.

## My Opinion on the Opinion of Chris “Monty” Montgomery

Chris Montgomery is the Audio Expert, who invented the OGG Vorbis codec. That gives enough reason to accredit him with good advice. I recommend that my readers read his advice here.

I did read the whole thing, but have three comments on it:

1. The Author suggests that 16-bit sample-depth offers a de-facto solution to the limits in dynamic range, simply due to the correct application of dithering. If I cannot trust my hardware to perform correct low-pass filtering, why on Earth would I trust it to perform correct, 16-bit, audio dithering?
2. The Author explains the famous loudness curves, that define threshold of perceptibility, as well as the higher threshold of pain. What he fails to point out is that these curves assume, that the sound being tested for, is the only sound being played over the headphones. If there is another, background sound being played – i.e. the current loudness-level already higher than zero – then the threshold of perception for a given test-sound, is higher – requires a higher level, for that test-sound itself to be heard. Yet, this level is still lower, than the peak level of the background sound. People who design codecs know this, as I am sure the author does. It is the threshold of perceptibility next to a background sound – not the absolute threshold – which gets used in the design of codecs.
3. The Author suggests it would be a misuse of his codec, to encode discrete multi-channel sound. And one reason he states, would be the waste in file-size, while the next reason he states, would be the fact that sound jumps to the nearest speaker, when they are all encoded that way.

This last observation strikes a cord with me. I have already noticed, that OGG Files do allow numerous channels to be encoded in parallel, but that if we exceed 2, we lose the benefits of Joint Stereo. By itself, this does not really count against this Author, whose codec therefore does not offer explicit surround-sound. But the possibility is very real, that the localization of sound will jump to the nearest speaker, if the listener moves and the sound was encoded that way. It is entirely possible, that purposeful encoding of surround-sound by the (competing) AC3 or the AAC codecs, reduces this risk.

But then I would suggest an alternative approach, to people who do not want to use the proprietary codecs, yet who wish to encode their movies with surround.

There exists the Steve Harris LADSPA plug-in library, which includes a matrix encoder for Pro Logic. This matrix encoder accepts 4 input channels, one of which is the surround channel, and outputs 2 stereo channels.

Further, the circuitry must exist someplace as well, to accept 2 stereo, 1 center and 1 surround-channel, and to encode those in real-time, so that the surround-effect can be played back over 6 speakers.

• In principle, it should be possible to OGG-compress 4 channels and not 6, so that these channels can be used as inputs, to a matrix surround-system, like to the LADSPA plug-in, so that listenable surround will emanate from all speakers. Does Audio Software exist, which applies the LADSPA plug-in in real-time?
• Alternatively, it might be possible to mix down Pro Logic sound into Stereo using the Steve Harris plug-in, and then to use FLAC on the resulting stereo.

BTW: What the Author mainly writes, is how incorrect it would be for pure listeners, to download their music in 24/192 format. He does not actually write, that Music / Sound Authors should avoid recording in this format. And so one fact which I have observed, is that there exists a lot of Audio Software – such as – that stores its sound in some higher, internal format, but which, when instructed to Export that to a 16-bit format, offer Dithering as an option.

This is possible because the Application is numeric and not physical. Thus, If I had used my USB-sound-device to record in 24-bit, I could next Export the finished sound tracks to 16-bit:

But, It would also seem that Chris Montgomery equates the use of such technology, as only being suited for Professionals. I am not a professional, and do not have the extremely expensive tools they do. Yet, I am able to author sound-projects.

Dirk

## I just custom-compiled Ardour 5.3.0

I know an acquaintance, whose name I will protect, who uses “Garage Band” on his Mac, but who has a hard time imagining that there exist many, many different programs like it, for other platforms, and that there must exist such, in order for professional musicians also to have access to a great array of such tools.

Of greater relevance is the fact, that such software exists under Linux as well – not just on Macs or PCs – as well as under Android.

And there is one observation which I would like to add, about what form this takes if users and artists wish to do audio work using Free, Open-Source applications.

Typically, we can access applications that do most of the work that polished, commercial examples offer. But one area in which the free applications do lag behind, is in the availability of sample packs – aka loops – which some artists will use to construct songs.

If Linux developers were to offer those, they would probably also need to ask for money.

Further, Garage Band has it as a specific advantage, that if such loops are simply dropped into the project, this program has the tempo stored, with which that loop was playing by default, in addition to which all DAWs have the tempo of the project set and available. Garage Band will automatically time-stretch the loop, to adapt to the project tempo. Most of the DAW programs I know, do not do this automatically.

A common ability the open-source applications offer though, is to time-stretch the sample manually after importing it, which can be as easy as shift-clicking on one of the edges of the sample and dragging it.

In order for this to be intuitive, it is helpful if the sample has first been processed with a Beat Slicer, so that the exact size of the rectangle will also snap into place with the timing marks on the project view, and the sample-tempo will match the project-tempo.