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In my bug report I posted some research about this...

Vista and higher changed the way process priority happens so that the window with focus gets a higher process priority.

This is something Microsoft provides an API for so that it can be changed or disabled. Commercial daws would have fixed this in the Vista days, however our devs don't run Windows at all so the users have been experiencing this wothout much resolve.

I can understand ASIO4ALL fixes some windows only issues and it is probably helping by taking away CPU resources and actually allowing both processes to coexist without as many resource conflicts.

I believe by disabling this feature we will have better VST performance, but unfortunately ASIO4ALL uses non-GPL libraries so we can't bundle it and we tend to recommend other avenues first.

I am not normally a c++ developer (rather a user that has taken an active roll in fixing bugs and helping with cross-platform compatibility) but I can and do make PC and Mac builds so I will take the steps necessary to try the aforementioned fixes and update the bug report accordingly. Any further research on this matter should go in that same bug report until the issue is resolved.

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In my experience increasing the thread priority in programs does increase performance but it also makes them a lot more unstable and more likely to crash

I get why ASIO4ALL isn't bundled but couldn't implementing the ability to use it be an option? Their's a variation of it for linux as well so adding the ability to use it may be beneficial to those who use windows vsts with linux as well

I know it's developed by linux users but increasing performance for windows and mac users would be beneficial to the project as most people(even linux users) use windows or macs for making music so their's many more potential contributors on those OS's
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Free samples, presets, etc.

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Hi again everyone,
running LMMS on another machine I normally don't use, I couldn't get things set up correctly. I couldn't get any sound; so I ran it in a terminal and got this list of messages:

Code: Select all

VST_PATH not set, defaulting to /home/bodhi32/vst:/usr/local/lib/vst:/usr/lib/vst
ALSA-sequencer: unhandled input event 66
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused

ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused

ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused

ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused

ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream
WARNING: gnome-keyring:: couldn't connect to: /tmp/keyring-ZSEYoJ/pkcs11: No such file or directory
unique ID: S1Vs
RemotePlugin::DebugMessage: inputs: 0  output: 2
RemotePlugin::DebugMessage: creating editor
RemotePlugin::DebugMessage: editor successfully created
RemotePlugin::DebugMessage: failed getting shared memory
I am using Jack, which it apparently isn't seeing, even though Jack was selected in the set-up page.
Any suggestions much appreciated!

brian
Tired of Windows? Linux offers hundreds of good distros. For more info:
DistroWatch
Some good synths for linux: www.linuxsynths.com

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Have you tried getting rid of PulseAudio on the other machine? I don't know if it still applies but when I used Linux for making music it was usually recommended to get rid of it. Based on the log messages you posted it seems like PulseAudio is interfering.

Another cause I can think of is a misconfigured .asoundrc file (due to the references to cards.pcm.rear, etc.).
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I checked about pulseaudio, but in fact I never installed it, so that can't be the problem.
I did the update today, and was happy to find some new features, such as monster. So I'm excited to get it running; unfortunately I cannot get it connected to Jack. (It connects, but Jack then stops producing sound and I have to restart Jack.)

Regarding the .asoundrc file, it might be the issue, but I don't know what should be changed. Any suggestions?

Somehow, I get the idea that there is a setting wrong in the preferences, but I don't know what. As an alternative to Jack, I selected Alsa as the audio host, and it worked that way. But I want to use Jack, not Alsa.

brian
Tired of Windows? Linux offers hundreds of good distros. For more info:
DistroWatch
Some good synths for linux: www.linuxsynths.com

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@briandc,

I'd love to help, but I've personally never set up Jack with LMMS before.

If the KVR boards don't get you anywhere, you may have more hits here:

http://lmms.io/forum/

And of course our bug tracker if you determine something is actually broken.

-Tres

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Thanks tresf! Will do...


brian
Tired of Windows? Linux offers hundreds of good distros. For more info:
DistroWatch
Some good synths for linux: www.linuxsynths.com

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@briandc: So it seems that you have Jack running. Have you already tried connecting to the Jack server with some other Jack enabled application? In case it is possible to connect with other applications this would indicate that it is likely a problem on LMMS' side.

I'd also try to replace the .asoundrc with a clean on. For example the one described here:
http://wiki.ubuntuusers.de/.asoundrc.
Feel free to post the content of your .asoundrc here.

Last but not least make sure that pulse audio is really not installed / running. For example try running any of the commands described here:
http://en.wikibooks.org/wiki/Configurin ... io/Testing
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I've done the PulseAudio testing, and I'm sure I don't have it installed. The asoundrc file, however, I cannot find. My guess is that on BodhiLinux the file is under a different name.
I have found that, if I connect an instrument to Jack, and it works fine; then I connect LMMS and try to load a VST in VeSTige, then as soon as it is loaded, the sound is gone from Jack, both in LMMS and the other instrument(s) that were working fine. The only solution is to restart QJackCtl completely. (I prefer QJackCtl to other setups, at least for now.)

I posted the issue on the LMMS forum too, and hoping to get a response there...


brian
Tired of Windows? Linux offers hundreds of good distros. For more info:
DistroWatch
Some good synths for linux: www.linuxsynths.com

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We're ramping up for the 1.1. release with a brand new promo vid...

https://www.youtube.com/watch?v=W6tEolVz3_4

Testers welcome. Download here: https://lmms.io/download


Oh yeah, and a brand new website. :)

(preview of new website below)
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Hi,
For windows users, I've seen a lot complain about latency issues.
I wrote a post on lmms's forum about this, but I'll also post it here, maybe it's more visible:
https://lmms.io/forum/viewtopic.php?f=7&t=1761&start=10#
Hi Guys,

The lag/delay between the time when you press a key and you hear a sound is called LATENCY.

Latency has *nothing* to do with MIDI, and nothing to do with the length of your cables.

Latency is given by the buffer size your audio program uses to talk to your soundcard, in order to produce sound.

Simply put:
latency [miliseconds] = buffer size [samples] / sampling frequency [kHz]

For example:
If buffer size is 256 samples, and sample frequency is 44.1 kHz then latency = 5.8ms = 256 / 44.1

Less than 10ms latency is good enough for realtime audio for keyboards. Drum players will notice a latency in the 3ms-5ms range.

In order to decrease latency, the only thing you can do is to lower your audio buffer size.
Now this is where the audio software, audio drivers and hardware comes in:

Your system has to be able to fill up the buffer fast enough, or you'll notice cracks and artifacts in the sound (caused by playback discontinuities).
This means the software application needs to be able to talk to the soundcard as fast as possible.
On Windows, the only platform specifically designed to provide real-time audio is ASIO!
That's why you should use software capable to talk to ASIO drivers, and have hardware with dedicated ASIO drivers.

For hardware which is not designed for music making / realtime audio, there is a hack driver called ASIO4ALL. This can have better or worse performance depending on your actual hardware.

As far as I know, LMMS for windows doesn't have native ASIO support!
However, the latest builds of portaudio do have ASIO support in them.
The solution to make LMMS to support ASIO is to replace a dll file from the location where LMMS was installed to a newer one.

Old dll, which has to be replaced is called (located probably in C:\Program Files\LMMS or equivalent):
libportaudio-2.dll

New dll, can be downloaded here, depending if you're running in 32-bit or 64bit:
https://github.com/spatialaudio/portaudio-binaries
Here you can download either:
libportaudio32bit.dll
or
libportaudio64bit.dll

You have to download the dll corresponding to your LMMS version (32 or 64bit) and overwrite this to the original libportaudio-2.dll

Now if you choose portaudio as Audio Interface in LMMS, you should have access to the ASIO drivers installed to your system.

From there, if you have a hardware dedicated ASIO driver (like for a USB interface) use that.
If you have a generic Realtek soundchip, then install ASIO4ALL, the latest version, and use that.

Lower your buffer size to the minimum so that you don't hear audio artefacts, and then you can be sure you have the lowest possible latency.
== VDX == One Man can make a difference!
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> As far as I know, LMMS for windows doesn't have native ASIO support!
However, the latest builds of portaudio do have ASIO support in them.
The solution to make LMMS to support ASIO is to replace a dll file from the location where LMMS was installed to a newer one.
We ship LMMS with PortAudio but there must be something different about it over the dll you've provided.

If you can help research and fix this upstream,it would be much appreciated. Here's the thread to discuss it:

https://github.com/LMMS/lmms/issues/1173

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Can't get it to recognise my midi in on Mac - also how do I load in ZynSub?

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aMUSEd wrote:Can't get it to recognise my midi in on Mac - also how do I load in ZynSub?
Midi on Mac will be introduced in 1.2.0. We have a beta version available on our downloads page if you'd like like to give it a test drive.

Zyn GUI is broken on Mac. This hasn't been fixed with 1.2.0, we need help on this.

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tresf wrote:We ship LMMS with PortAudio but there must be something different about it over the dll you've provided.
If you can help research and fix this upstream,it would be much appreciated. Here's the thread to discuss it:
https://github.com/LMMS/lmms/issues/1173
I'm not affiliated with the guys who compiled these dlls. I just found out that they work, and do have ASIO support! (both 32bit and 64bit!!)
Last edited by jackoo on Sun Mar 13, 2016 1:45 pm, edited 1 time in total.

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