Hey hey,
I have just released "A Moment of Passing", which my friend Ritual Notes and I
wrote and performed together. The recording is a few months old, but I just
got back my production environment.
https://youtu.be/dmTR8Rm70sY
My friend plays the flute, I play the piano. We did record it in two goes,
since I don't have two headphones or anything.
It's a sad little song, with a few sunnier moments and it came to be on the
spur of the moment. There will be another song, which was properly composed,
in two days time.
Enjoy and best wishes,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
Make all the clouds disappear
Put all your fears to rest <3
(Britney Spears)
aloop is a audio file looper for Linux using PortAudio as back-end
(jack, pulse, alsa), libsndfile to load sound files and zita-resampler
to resample the files when needed. The GUI is created with libxputty.
This is the first release for aloop, it comes with the following features:
* support all file formats supported by libsndfile.
* resample files on load to match session Sample Rate
* file loading by drag n' drop
* included file browser
* open file directly in a desktop file browser
* open file on command-line
* create, sort, save and load playlists
* select to loop over a single file or over the play list
* move play-head to mouse position in wave view
* set loop points for start/end loop
* save loop points in play list
* save selected loop as wav file
* play backwards
* volume control
* break playback (keyboard support space bar)
* reset play-head to start position (keyboard support courser left)
Dependencies
* libsndfile1-dev
* portaudio19-dev
* libcairo2-dev
* libx11-dev
Project page:
https://github.com/brummer10/aloop
Release Page:
https://github.com/brummer10/aloop/releases/tag/v0.1
Please report issues to the project issue tracker.
Hi!
This mail is just a heads-up about a finding discovered with the help of
the linux-usb mailing list and which I thought some other people might
benefit from. If this is nothing new to you, then please do ignore it.
Background:
The USB audio class 2.0 specification dictates that isochronous
transfers (i.e. audio frames to/from an audio interface) happen every
"micro-frame". USB micro-frames are 125 microseconds (us) apart. 125 us
= 0.125 milliseconds (ms).
The majority of USB audio interfaces (at least those that I have) use
synchronous audio-streaming, i.e. the sample clock is derived from the
bus clock. There are definitely interfaces that use e.g. adaptive or
asynchronous modes and this discussion would have to be altered for these.
Given the case of synchronous mode isochronous transfers at a sampling
rate of 48000 Hz (= 48 kHz) this would correspond to 6 audio frames per
USB micro-frame.
48000 frames/second * 0.000125 seconds = 6 frames
So in principle a very well behaved audio interface attached to a very
well behaved USB controller sitting in a well tuned system _should_ be
able to achieve a minimum round-trip latency of 2 * 6 frames = 12 frames
or 250 us. This leaves out additional buffering inside the audio
interface and additional latency by anti-aliasing and reconstruction
filters.
The first caveat to above in the context of Linux: The snd_usb_audio
driver does not seem to support period sizes of just 6 frames. It _does
support 12 frames though, which is nice.
And here is the other caveat which lead to the title of this mail: Some
controllers are behaving "worse" than others. There's this little thing
in the XHCI spec which amounts to the following: The XHCI can specify in
a register how many micro-frames have to buffered at all times for
outgoing isochronous endpoints. The Intel XHCI in my ASRock N100dc-itx
main-board for example request a whole USB frame (which corresponds to 8
micro-frames or 1 ms) to be buffered at all times. Another XHCI that I
have (a Renesas controller) only requires one micro-frame. This has
direct consequences on the kind of period sizes and number of periods
that are usable on these controllers. These consequences don't explain
everything but at least you know that you can't ever get better than
this limit.
For example for a period size of 48 frames I need to use 3 periods on
the Intel controller (resulting in 3 ms round-trip latency), but for the
Renesas I can use 2 periods at 48 frames (resulting in 2 ms round-trip
latency). On the Renesas controller even 2 periods at 24 frames works
fine (1 ms round-trip latency).
I can lower the latency on the Intel controller by using a smaller
period size but more of them as long as the buffering requirement of the
controller is satisfied. One stable setting is for example a period size
of 24 and 5 periods which results in a round-trip latency of 2.5 ms.
So what's the take-away here? In some cases, if you are chasing stable
low-latency operation using a USB audio class 2.0 device it might just
be worth installing a different XHCI in your computer (the above
mentioned Renesas controller is just a PCI-Express card which sits in a
slot in my N100DC-itx board) that is better behaved than the one you
currently have.
Another take-away is that the above limitation only applies to the
outgoing direction (playback). If all I was interested in would be the
capture direction then 2 periods at 48 frames would work fine even on
the Intel controller.
Kind regards,
FPS
Hi list,
Since a while (recent kernel or other package upgrades?) I am
experiencing serious problems with jackd after resuming from suspend.
Eg. trying to play back audio files via mpv yields
mpv somefile.flac (+) Audio --aid=1 (flac 2ch 44100Hz)
Cannot read socket fd = 13 err = Success
CheckRes error
JackSocketClientChannel read fail
Cannot open mpv client
JackShmReadWritePtr1::~JackShmReadWritePtr1 - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
At which point the command hangs. The card used is an Edirol UA-25
connected via usb.
I can quit the hanging mpv with Ctl-c, upon another invocation the error
persisits but mpv gracefully attempts alsa this time.
Qjackctl, which was used to start jackd in the first hand place, is
unresponsive as well. I for now kill jackd as root three times to regain
control over it after every resume.
How can I go about to debug this further?
The above problem is with jackd2. Switching to jackd1 will show a
slightly different problem upon resume: Using 100% CPU and being
unresponsive either. I posted about this in jan/feb 2024
By the way: I can't search in the mailing list archives as the search
engine on linuxaudio.org throws a lot of "502 bad gateway" or "504
Gateway Time-out" errors at least for lac and jackd lists:
https://lists.linuxaudio.org/hyperkitty/search?mlist=linux-audio-user%40lis…
Thanks!
Peter
Hello,
On 1/20/25 18:11, Peter P. wrote:
> By the way: I can't search in the mailing list archives as the search
> engine on linuxaudio.org throws a lot of "502 bad gateway" or "504
> Gateway Time-out" errors at least for lac and jackd lists:
> https://lists.linuxaudio.org/hyperkitty/search?mlist=linux-audio-
> user%40lists.linuxaudio.org&q=resume
This is because the mail.linuxaudio.org server is a bit under-powered.
And it's under-powered to keep the costs low. If people are (still)
willing to donate then I should offer an option to do so I think, that
might help increasing the performance of the servers a bit. What is the
best option for receiving donations these days?
Best,
Jeremy
