What steps will reproduce the problem?

1. Install Soundflower 1.4.3 with Soundflowerbed
2. Use (2ch) as Main Output, so I can use the Volume controls on my Audio DJ 8
3. In Audio Setup, I use 96kHz and 32 Bit (well actually the Soundcard does only 24 bits, but there is only this choice) It does not seem to matter which channel is used on the Audio Hardware.

What is the expected output? What do you see instead?

After a while, the Sound is completely muted or I can hear some strange humming distorted nois which sound like a Buffer Overrun. In fact, with a smaller Buffer, the Artifacts can be heard sooner and generally with a higher pitch. For example, with a Buffer Size of "2048" it takes about one hour before the Artefacts appear (seems to be a fixed amount of time), with a buffer size of "64" its only a matter of minutes.

All in all, this Bug reminds me of "Issue 21" - so this might not be related to my Audio Hardware

I can "fix" the Bug by just selecting any Buffer Size when the error appears, but of course it will reappear a certain amount of time (depending again on Buffersize).

What version of the product are you using? On what operating system?

I use Soundflower 1.4.3 in Mac OS X 10.5

Please provide any additional information below.

Audio Hardware: Audio DJ 8 from Native Instruments

I have the exact same problem, but my Interface is a Native Instruments Audio 4 DJ.

I've also tried the latest drivers from Native Instruments, without success.

I have the same issue with a MOTU Ultralite on 10.5.7

I'm using Mac OS-X Leopard 10.5.{6,7} also.

Any hints on how to solve this bug?!

I'm running the latest version of Soundflower on Snow Leopard (MacBook 3.1), with latest drivers.

It appears to be some kind of buffer overflow. When the artifacts occurs, clicking on Soundflower menu item and reselecting the interface temporarily solves the problem.

I also have an issue just like thi.

SF/SFB worked perfect on Leopard, I am wondering whether to go back until a fix is found? As this seems to have happened sine i upgraded to SL. Although I have not confirmed that yet...?

Ableton 8.0.05 into SF 1.5.1 then into Audacity 1.2.5 // AH Pro 2.9.4

Have also tried through Audio HiJack, but, because the trial is time locked, 10 minutes is not enough to see if the problem replicates through that app...?

Forgot to add the specs...

MacBookPro 5.2 17 Uni Intel Core 2 Duo 2.66 GHz 4 GB

Snow Leopard 10.6.1

This bug occured with Leopard too. I've never been able to suscessfully run Soundflower with my NI Audio 4 DJ sound car.

Issue 21 has been merged into this issue.

Issue 19 has been merged into this issue.

Issue 10 has been merged into this issue.

Issue 40 has been merged into this issue.

I have reproduced this problem using Max/MSP generating a sine wave with the AudioTester patcher -> SoundflowerBed.

Comment deleted

Comment deleted

This problem does not require a Firewire audio interface -- it will occur with the built-in sound device.

It is not at all obvious from looking through the code where the problem is occuring.

MacMini Intel Core Duo 1.66 GHz 1GB 10.6.1

Do you have an estimate on when this bug will be solved and a new version released? Thanks.

Phew, I'm glad this has been picked up.

I am gutted not being able to lay anything down at present, without sitting on tender hooks, waiting for the drops and clicks.

If you need any audio samples, feel free, I can provide to demo the problem

Glad these apparently related issues have been picked up & merged. Thank you - please mail me if there is anything specific you would like tested.

Coming in with my 2¢ here. I'm running soundflower on snow leopard, on a macbook3,1 without any issues.

One thing I've noticed about my laptop is that it ALWAYS is emitting faint buzzing and 

higher pitched beeping, whenever the soundcard is powered on. This is not a soundflower issue; this is a hardware issue. It is only noticeable when using headphones and when no other audio is playing (but the soundcard has not been cycled off). Additionally, messing with the audio settings, such as switching audio devices, will cause the soundcard to cycle off and "fix" this issue.

Could it be possible that some you with the "buffer overrun" issue are simply experiencing 

the effects of the macbook's relatively average soundcard as well? Of course, this doesn't apply to those who are experiencing high cpu usage as well, I believe that's a separate issue.

@percontation: no, this has nothing to do with what you described. 1) if you hear the sound you will immediatly know it is not an hardware issue 2) it happens on certain Soundcards, not with the apple chip. 3) I don't have a macbook ;-)

So really, it is soundflower itself which needs to be fixed

@percontation: I have a NI Audio 4 DJ interface and a MacBook 3,1. It does emit the faint buzz ONLY when my MacBook sleep or is powering off. When it's on, the faint buzz does not appear.

This faint buzz was introduced since I upgraded my system to Snow Leopard. I don't know if it's a software issue, or if it's just a coincident hardware failure.

I don't know if it's related to SoundFlower also. But I have completely uninstalled SoundFlower and the faint buzz becomes evident when I put my MacBook to sleep.

But this is a issue that doesn't botter me at all.

@percontation: What does botter me is that SoundFlower does not work. Whenever I run it with my Native Instruments Audio 4 DJ, after some period of time, the audio is "jammed", and the weird sound artifacts appear. The audio only returns to normal when I click in some option of the SoundFlowerBed icon in the menu bar.

* But I think this bug may be perceived ONLY with expensive audio interfaces with very low latency. Cheap ones, like the onboard audio interface of the MacBook, may not have present the issue. *

Evidently these are different issues indeed. It just struck me how much your textual descriptions would resemble me trying to describe the crappy mbp built-in audio. I guess I'll fiddle around with the two USB audio cards I have and see if I can hit the problems you've been talking about (I've used them before without problems, though).

I Have the exact same issue with a native instruments audio kontrol 1. After a few (less buffer) or more minutes (more buffer) the audio either stops completely (happens with vlc video) or is completely jammed (using itunes/front row). Would it be possible as a workaround to inlclude in a new version of SF an option with a much bigger buffer ?

Comment deleted

Disagree with comment 26 - not a true representation of the scope of the issue.

Sorry, @shoarthing, it's just a guess.

This problem also occurs on a hackintosh (OS X Leopard, a Gigabyte GA-P35-DS4 motherboard which has Realtek ALC889A integrated audio).

I used Soundflower to enable audio volume control when using optical output. But I stopped using it because of this bug. Hopefully this will get fixed soon and I can use those volume buttons on my keyboard again :)

Comment deleted

Unfortunately, I gave up on waiting.

When i reinstalled Leopard on a different partition the same issue appeared, which then indicated mine was hardware.

So, following a board replacement, I have actually found a work around with Ableton, record the audio in there and then render to Wav, you don't need SF then.... Da Dah!

Any position about this bug? Thanks.

Please fix this fast! These Problem makes Soundflower absolutely not usable at all for now.

yes, fix it! I reported this over 6 months ago and a LOT of people can't use this. This is even more serious bc Soundflower remains the only option for certain situations. PLEASE, fix this ASAP. Thank you!

exact same issue here. selecting 2048 buffer size gives about an hour before it gets "jammed", and clicking any current menu setting SF seems to 'reset the clock' as it were on the countdown before it jams up again. i use (or would like to use) SF for streaming audio from my computer and would like to leave it to do its thing automatically without coming back every hour to click on the SF menu to reset the "jam countdown" (what i would guess, as others have guessed, is a buffer underrun issue).

until it's fixed i'll try to come up with an applescript that essentially does that menu click, or equivelant, for me, every hour.

crossing fingers that this gets fixed soon, cos it's a great and quite necessary app/function!

would be glad to do testing.

10.5.7 3 gigs ram onboard audio

I guess there is no one looking at this bug (or this messages anyway...)

we are working with final cut studio and are facing the same problem. soundflowerbed seems the only option to us....since months we are working with this horrible bug, it's no longer possible for us...

is there any solution upcoming?

I don't know why this has started happening! I was using Soundflower to route all apps out of my M-Audio Projectmix's digital output so I can use the much better D/A conversion on my MOTU Traveller.

It was working great for many months, but when I upgraded both Ableton LIve and MAX/MSP, I started getting these weird buffer artifacts, which I wrongly thought for a while were related to those programs updates, but now I get them eventually on any audio source when using Soundflower.

This clearly didn't happen before, but now it does. I am confused.

Just wanted to add that I have noticed the same problem, confirm the same workarounds as effective to solve it too. Platform is Snow Leopard 10.6 along with Soundflower 1.5.1, runnning in 32-bit kernel mode. Soundcard is onboard (Maximus Formula) with a Analog Devices chip AD1988B running an unsupported kext. I know this may not the best controlled environment you get but there it is. I may just test 64-bit to check if there's any influence. I'll report later.

Thank you, @gamma.stardust. Please, do you have any idea of when a new version (or beta) will be released solving this?

@filipead, nopes... have no idea, I am just a newcomer around.

I am testing 64-bit right now... I'll report soon.

Test done, 64-bit suffers the same issue. This was made with a buffer of 512. With Soundflower 16ch output of a 5.1 ac3 track played on VLC 1.0.3 (32bit)

Following up with a minor report.

Same stuff happens with Quicktime Pro X.

Same problem here, using an iMac 2007 intel aluminum 24 inch with Snow Leopard 10.6.2 and a MOTU 896. I have all sound output routed to soundflower 2ch and soundflowerbed 2ch routed to my Motu 896 to control volume via software. Works great for maybe half an hour and then the sound starts to sound robotic and then locks up looping on a sample. Changing output drivers fixed it for me.

Same prolem with a M-Audio Fast Track Ultra (FTU) on a 2.66 GHz Quad with 10.5.8. ...

same on dual quad 2.8 intel mac pro, imac alu and mac book pro. all running 10.6.2 and the problem is there when audio is routed from inside the computer to an external soundcard. happens with a MOTU 828mk3 firewire audio, MOTU ultralite mk2 (both MOTU drivers) and EDIROL fa66 (coreaudio drivers). sound is ok for some 40-50 seconds and then KCHCHCHCHCHCHCHCH. noise. switching the outputs (resetting the clocks?) makes things right again for a moment. it happens on 32 and 64 bit startup modes.

the first time this happened was with right after installing 10.5. with 10.4 it worked fine. I was hoping for it to go away with 10.6 but no, something is still screwed up and looking over the code, no obvious fix is seen.

I really wish this could work again someday in a not to far future.

It appears that Soundflower is an abandoned project... This bug is confirmed by several people, but no developer is trying to fix it. Too bad, it's a great tool. I would even pay for it to work properly.

Issue 44 has been merged into this issue.

Think Issue 43 is also describing the same problem. So, can we hope for a fix soon? Seems that this is issue #1 for many many people. I hope you will have the ressources to do this. And, as others I would even pay for a fix! Thank you.

Issue 43 has been merged into this issue.

Trying to look into the source of this problem and why affects some users but not others...

Most of the comment and reports of this problem are missing information. In order to be helpful they need to include, at a minimum, the following:

1. The signal path. e.g. Live 5.1.1 -> SoundFlower -> Max/MSP 5.1.3 -> built-in output

2. The sample rates and vector/buffer/block sizes used by all programs in the signal path

3. Hardware configuration (many have included this, which is a good start)

4. Exact and detailed steps on how to reproduce. Vague instructions or those missing steps don't help.

Thanks everyone for your help! Once the problem can be pinned down then there is hope of trying to fix it.

Having now spent a couple of days on this problem, I am only able to reproduce when I use a bogus setup. I am unable to reproduce when everything is set up properly:

• all involved applications are using the same driver (e.g. Built-in or Apogee Duet or whatever) and not mixed drivers
• all involved applications are running at the same sample rate
• all involved applications are using the same vector size
• the vector size is reasonable (i.e. 512 is reasonable, 32 is not)

This is with a beta build of Soundflower 1.5.2, so it's possible that some of the issues involved are also solved in-part by this new version, which I'll post shortly.

Tried with Soundflower 1.5.2 beta and the issue appears in it too.

The signal path is VLCPlayer -> SoundFlower -> Native Instruments Audio 2 DJ Buffer size is 512 or 1024 (with 1024 it takes more time to reproduce)

MacBook MB404RS/A

It's quite easy to reproduce the problem, just listen to music for 5 minutes.

BTW, I'm familiar to programming, so I can help you collect logs if you'd like this.

Forgot about OS: Snow Leopard 10.6.2

ok. tested again.

10.6.3 latest developer build, stared up in 64 bit mode soundflower 1.5.2b1 early 2008 mac pro 2 x 2,8 GHz Quad-Core Intel Xeon, 8GB DDR2, 6 GB striped raid firewire audio with Motu 828MK3 and Motu UltraLite

simple audiopath 1 : iTunes > soundflower 2CH 1 + 2 > Motu 828MK3 CH 13 + 14 (SPDIF)

simple audiopath 2 : Soundtrack > soundflower 2CH 1 + 2 > Motu Ultralite CH 5 + 6 (ANALOG)

all set to 48 kHz, soundflower buffer 512

it takes longer to reproduce with bigger buffer, but it will happen. it also happens if I select any other combination of channels on the output devices.

after 12-13 minutes (longer than ever before, some progress at least) : audio stalls in last frequencies, providing an almost continuous tone.

good luck, we all have our fingers crossed that this might get fixed eventually.

10.6.2 (32bit), MacPro 2.66 Dual Listening to iTunes, Safari, various. (normal use) Stopwatch timed, not accurate.

system audio settings: Use This Device for Sound Output / Input = Soundflower (2ch-32bit) @44.1kHz->EchoAudioFIre(12ch-24bit) @44.1kHz, clock source = Mac

buffer: 256 = 3:25 to meltdown buffer: 512 = 6:39 to meltdown buffer: 1024 = 13:15 to meltdown

system audio settings: Use This Device for Sound Output / Input = Soundflower (2ch-32bit) @44.1kHz->EchoAudioFIre(12ch-24bit) @96kHz, clock source = Mac

buffer: 256 = 1:59 to meltdown buffer: 512 = 3:09 to meltdown buffer: 1024 = 6:15 to meltdown

system audio settings: Use This Device for Sound Output / Input = Soundflower (2ch-32bit) @96kHz->EchoAudioFIre(12ch-24bit) @44.1kHz, clock source = Mac

buffer: 256 = 1:42 to meltdown buffer: 512 = 3:10 to meltdown buffer: 1024 = 6:12 to meltdown

let me know what other factors to compare, t. no problem consistently reproducing it for ya.

These last two comments don't indicate what software is getting the sound out of Soundflower and sending to the hardware, which seems to imply that you are using SoundflowerBed?

I just tried SoundflowerBed and I can reproduce the problem. So this looks like a bug not in Soundflower, but in SoundflowerBed. Soundflower itself seems to work fine with other applications (such as Max/MSP) at the end of the chain.

I too am suffering from this problem. And Yes I am using soundflowerbed to route the audio to my audio interface (Mbox 2 USB).

I am running OSX 10.6.2 , Mac Pro.

Signal flow is Logic Pro 9.1 - soundflower - Soundflowerbed - Mbox 2 output.

Buffer is set to 512 samples in all applications.

I get about 5 minutes or so before I have to reselect my hardware from soundflowerbed to get sound again. This happens every time I use soundflower 2ch or 16ch as the audio interface for Logic Pro.

Also, when I first start soundflowerbed although my audio interface is already selected in soundflowerbed it does not route audio to it until I select the interface again.

Thanks

Yes. Using soundflowerbed. I will try with max tonight.

I was also using SFB....

Although have given up on this software for now and have found a work around for my particular set up.

using previous config, the problem exists when using SFB, not with MAX.

I have the problem with the following signal path: Traktor Pro => Soundflower => Ableton Live 8 => Audio8 DJ. I do not have the problem with the following signal paths: (i) Traktor Pro => Audio8 DJ (ii) Ableton Live 8 => Audio8 DJ (iii) Traktor Pro => Soundflower => Live 8 => Internal Sound Card

It appears that something about my configuration that involves Soundflower finally routed through the Audio8 DJ driver seems to cause this issue.

I have a similar setup like vamshi.raghu, and I also have exactly the same behaviour like all people here reported (noise after certain periods of time). Might add that I am also using Soundflower, so the Setup

Anyapp => Soundflower => Audio DJ 8

produces the problems stated above by me and other people. Should be really easy to reproduce and it also seems to occur on certain types of sound hardware, apparently including all Native Instruments audio interfaces, but also many others. Maybe all of these cards have a similar chipset?

I'm really happy to see some progress on this issue, and also come cleanup/consolidation of the reports.

I'll try to re-verify on my current setup (Mac Pro with Apogee Ensemble) ASAP, and verify whether it happens just with SoundflowerBed involved, or not.

The same bug occurs in the new Soundflower 1.5.2 Beta1. Too sad...

Soundflower was awesome when it worked. I was able to control the System Volume from my computer while using a firewire device through Soundflowerbed. Could we get the priority escalated from medium please?

-R

I'm currently working on fixing this issue myself. I seem to have narrowed down where the problem is occurring. If any of the maintainers have any input to my observations, that would be helpful.

In the InputIOProc of AudioThruEngine, input audio is stored in the ring buffer, and in the OutputIOProc, it is fetched. After a certain period of time, every attempt to fetch from the ring buffer in the OutputIOProc fails because the times passed in fall outside of the range of the buffer. So say the buffer is 96000 samples big (48khz / sec @ 2 seconds). On my sound card, the safety offset is 50, so it will consistently fail trying to fetch 95950 - 96462 (462 greater than the buffer) when buffer size is set to 512. Not quite sure what is causing this, but it is pretty consistent and has to do with the ring buffer. If I change the source to bypass the ring buffer, writing directly to This->mWorkBuf in the InputIOProc and reading directly from This->mWorkBuf in the OutputIOProc, the problem disappears, though there is a bit more static than normal because there is no safety offset.

Would any of the maintainers care to comment? I'll submit the patch if the solution is found.

I normalized the buffer times, and this is the error I'm getting from AudioRingBuffer::Fetch():

error - buffer times: 0 - 96000, reading for 95987 - 96051 This same error repeated > 1000 times

Thanks for working on this! Please do submit a patch if/once you get a solution working. I'd be more than happy to build up new installers as soon as this issue is resolved.

I'm also glad to hear there is progress in...and hope this issue will be solved soon...If we we can help with any further information, let us know...

just to add my stats:

Mac Pro Quad Core, 6 GB 1066 MHz DDR3, Mac OS X 10.5.8

audio path 1: Final cut Pro -> soundflowerbed (1.5.1, 2ch) -> Blackmagic Audio audio path 2: Final cut Pro -> soundflowerbed -> Level Meter (Spectre)

Buffer size is 2048, not editable in FCP, Signal is 48 KHz, 32 Bit, 2 Ch

I've been working on this for a few days now. Here's what I found out.

0) There are quite a few simple typo bugs in SoundflowerBed, those are trivial to fix. SIntXX, in audio app, there really isn't a lot of call of Signed Numbers (unless they are PCM data in float) I changed all that to UIntXX There in [AppController readDevicePrefs], val should really be unsigned long [AppController bufferSizeChanged16ch], should never use data from 2ch

1) The fundamental problem with this issue (24) is that you have two devices running in parallels without any effort to keep the two clocks in sync (beyond setting the sample rate the same). Which, in theory is enough, but in practice those two clocks can drift apart, especially if one of them is driven by different hardware than other.

Solution to this is to either: a) Keep track of drift between the two clocks and compensate by either re-sampling the data on the fly (hard) or insert some noise by duplicating/dropping sample frames (easy, but bad). b) Find a way to sync those clocks. But without actually having sync source that you can physically tie two hardware together, this might not be possible. c) Rewrite AudioThruEngine to use AudioQueue

I am trying to do c). I will post again if I get anywhere with it.

Thank you very much, @Jyin.Spheal. We all appreciate your effort.

Jyin.Spheal,

Can you release the version of Soundflowerbed with the typo bugs fixed?

-R

Comment deleted

I don't have write access to SVN. Besides, my code-base is currently a utter mess.

AudioQueue is a bust... not really designed for this sort of thing. Latency on AQ is in the order of 2-300 ms in i7 system.

Moving off to Plan d)....

what was plan d? i only saw up to c :) i've been working on your plan a & b, as I believe it's really the best way to implement it to give the most versatility. I've identified the exact problem and understand how it's drifting apart, i'm carefully considering the solution.

i'd be interested to see the code you've implemented for audio queue services, perhaps you could email what you have?

bryanmatteson hotmail com

The code I have for AQ is not functional, it is just functional enough to measure latency. As to Plan D... I am trying out AggregateDevice.

Basically, I will tie the input and output device into single aggregate device and use one IO proc to do the input/output in single call.

So, Plan D seems to be coming along well. I got prototype running on my system as I type.

I've just applied to this group, hopefully, by the time I get some testing done on this code-base, I will have write access to SVN.

I've ran into a problem I can't seem to fix. Every 30sec or so (with 2048 buffer) I get a pop in the sound.

Only on 2048 buffer? Or is this the case on, say, a 64 buffer as well?

64 buffer is unusable in my system (10 channel output). 2048 seems to give best result in terms of sound quality.

Comment deleted

Awesome to see some progress on this! As for getting it into SVN, there are a variety of options. The easiest way is if you are using Github @ http://github.com/tap/Soundflower . Usually the bleeding edge is there and the stable changes are then merged to Googlecode's SVN.

Otherwise we could get you svn access on googlecode and maybe this could go in a branch initially and then we can merge it to trunk after testing/review? To go this route, drop me an email.

Thanks, I will look into the qithub, the noise issue is still there, and I got bit busy with work at the moment. I will get back to it when things quiets down.

Any news, folks?

Any news, folks?

I have a similar issue - I get messed up audio buffers (ie a sine wave looks chopped up and rearranged - see screenshot) after ~ 5-10 min. I am not using SFBed's monitoring feature, but instead I'm aggregating Soundflower and my MOTU Ultralite. Everything has 128 sample buffers. Ultralite is clock source, resample not checked on either. Would love to see this fixed! For now I am using JackOSX instead.

Has anything else been done on this issue? I used to yse SF and SFB about two years ago and never had a problem like this one. Today I needed to use it for a project and can up with the same issues everyone above has had. My Setup:

Mac Pro 3,1, 8 GB Dual quad Routing: Kore 1 controller out 1/2 mapped to SF ch 1/2===> SF ====> NI Maschine input 1/2 mapped to SF input 1/2 with soundflower bed ch1/2 mapped to Kore1 controller out 3/4 (headphone outputs). Maschine external output mapped to SF ch2 which is mapped in soindflower bed to ch1/2 on the Kore1 controller AI

Whenever going into the audio menu of either app the sound would stop. If playing a sequence on maschine the sound would turn into a nasty buzzing simialr to an audio loop). Saw above manywre having problems with NI audio ints, so I switched the device to my Alesis USB 16.

Pretty much got the same results except no buzzing. The sound would no longer play throughthe AI, but I did confirm that SF will routing the audio cleanly. The audio routed from Kore into Maschine was showing on the Maschines meters. I hit record and got a waveform when done. Reset SFB and played back the sampled wav and it was clean, so it appears that the SFB connection to the AI was disonnected/dropped/hung?? Yet the audio channel between the apps was still good.... not useable, but contributing my experiences.

Sorry, I've been tied up with other project and haven't had a lot of time to work on this. I got aggregate code working, but I still get noise.

There may be some thing in the driver itself.

If I find time, I will insert some NSLog and see what I can find.

Any news, folks?

Is SoundFlower dead? C'mon, guys!

Yeah - soundflower is such a great concept and so useful and simple, but I've had to stop using it because of this issue. If I knew what I was doing in this realm, I'd help out.

Does anyone knows any alternative to this otherwise great piece of software ? I've tested Jack OSX but experienced the same issue... Soundfloweer is way more simple to use and configure. Would love to see it work again.

in my experience jackOSX works perfectly fine if you set it up right - its just that it is more complex, requires remembering to turn it on before launching your software (doh!), the connections GUI is not very good, and it seems to eat CPU for breakfast. The last is the reason I don't use it when performing - i get pops and clicks in the audio doing much less audio processing than I would if i was not using jack. For now I've been getting around this by using hardware loopback in my MOTU ultralite, but its not ideal - only 4 chans (2 chans in cuemix + 2 chans SPDIF loopback) and it means I am glued to that hardware.

Issue 79 has been merged into this issue.

same issue :-( after six hour with nrv10 and mac mini.

This issue is over a year old. What's the status?

The status seems to be "We simply do not care."

Has anyone found another solution yet?

I'm willing to PAY someone to actually fix this. Don't have huge amounts of cash to spare, but something. I know first hand how hard this stuff is to debug when it's in realtime and requires time before the bug manifests… I would work on it myself, I just know nothing about coreaudio and not sure i want to spend the time learning it. Or maybe Apple has fundamentally broken something and a fix is just not possible? (that would not be particularly unusual as I understand it)

Add me to the list of people that are willing to pay someone to actually fix this. "Don't have huge amounts of cash tos pare, but something."

I don't have lots of money, but I'm willing to pay for this also! Right now I could pay $ 100.

Please, SoundFlower developers, charge any amount for soundflower, but make something decent that really works!

/sign

I would also be paying for a fix ....

I believe fixing this issue involves the use of the kAudioUnitSubType_Varispeed AudioUnit, in case anyone is interested in pursuing it...

I have the same problem when a play a flac 6 channels 24 bits 96Khz. But when a play another flac : 2 channels i haven't artefact

Wow, I was really on the edge of my seat reading through the comments hoping that someone was going to come up with a patch for this. Not sure I could provide the detail necessary to satisfy a project member with my situation but I will say:

TONS OF MUSICIANS ARE TRYING TO FIGURE OUT HOW TO GET AI (ie MOTU) CLEAN SIGNAL ONTO QT VIDEO THAT INCLUDES AN AUDIO ACCOMPANIMENT (ie, backing tracks) RUNNING ON THE DESKTOP.

I am playing my bass guitar along with Band-In-A-Box and shooting iSight video into Quicktime X (the new dumbed down release...lol).

• MacBk Pro 2.4Ghz, Snow Leopard 6.2
• Aggregate Device Soundflower (1.5.2) + MOTU Mk3 Hybrid (USB for now)
• Bass ( and sometimes lav mic) ===> MOTU w/48k sample rate
• Quicktime X iSight vid window w/Aggregate device selected for sound
• Band-In-A-Box runs on the desktop (BIAB)
• Sound Prefs ===> input and output set to Soundflower

Problem 1) I have been having the same problem with the static that turns into a high pitched hum after some minutes of operation. (HOLY sheiss, Update......see below).

Problem 2) Took me a while to realize this, but since I was only soloing on the accompaniment, I did not realize that the bass signal had a slight delay on it, maybe 200ms or so. When I cut a groove track the other day, I thought: "does my groove suck that bad?...answer: no). I suppose that sending the MOTU through Soundflower puts in the delay..not sure.

UPDATE: Problem "1" solved by selecting "Soundflower" as Clock Source for the MOTU in Aggregate Device setup window.

SECOND UPDATE: Problem "2" Bass sound delay solved by lowering buffer size in SFB menu to "64".....I am going to do further research on this to the best of my ability. But the delay is gone and the groove is now back to smokin'.

The object has been to get the accompaniment from the BIAB to appear in the MOTU through the USB so I can shoot various video projects, such as instructional or gear reviews, etc, but to also include some high-quality accompaniment from the BIAB running on the desktop.

So Soundflower should be a great thing for this. I basically have figured this out for the most part, but like others after a few minutes the static would pop in and then I would have to reset (re-sync clocks?) from the SFB menu to get rolling again. In fact, as I am typing, I tried something different and IT"S WORKING!!!!!!!! ...........I have the BIAB rolling, a QT vid window open, and some setup windows open to try some adjustments stuff. I went into the Audio Device setup and on the MOTU, decided to select "Soundflower" as the Clock Source instead of "Internal" and UNREAL, the sucker isn't tweaking out on me now. When the MOTU has "Internal" set as clock source, the sound flakes in 4 minutes flat (with a buffer of 64 (to solve delay issue, but WAS 512 before)). It has been rolling with "Soundflower" as Clock Source for 24 minutes with no flaking.....UNREAL, I have reached the promised land after months of tech torture.

Here is a Soundflower/MOTU/QT/BIAB video I shot between static blowouts the other day after problem 2 was "solved" (temporarily...? don't know yet for sure):

http://www.youtube.com/watch?v=OeBP7ek9hwg

Solved the problem this morning "accidentally" by setting the source clock in "MOTU Audio Setup" to Soundflower. Since then (11 hours) no soundbreak or distortions or anything else.

Deleted Soundflower completely from the System (10.6.6) and reinstalled Soundflower 1.5.2.

huh… will have to test this! Never thought to do it that way.

People with different audio interfaces that don't have the ability to run off of another interface's clock, or people trying to use the built-in sound, may not be so lucky with this workaround. I'm curious what the clock jitter is like when the MOTU is synced to a "software only" audio interface such as SoundFlower…

Your solution to "problem 2" of course makes sense - I do all my performance stuff at 128 samples I/O buffer - a good trade off i find between latency and processing power. The smaller you set the buffer the less latency, but the computer has to work harder to do the audio processing because it has to process more smaller chunks instead of fewer bigger ones, and one of the things that actually takes the CPU a lot of time is the context switching between doing one process to the audio and doing another and jumping around in the memory (as opposed to stepping straight through). So fewer bigger chunks = more efficient, but because the beginning of the chunk cannot go out of the computer until the end has come in and the processing has been done, it results in more delay from input to output. Of course there is double or quadruple buffering thrown in there too to make it all work smoothly…

. . @Comment 95 by arvi...@gmail.com, Oct 08, 2010

Yes, I had high hopes for JackOSX, but I couldn't get past the setup process. I just don't understand all of the setup issues to be able to think it out correctly. If it does eat CPU time maybe it would not have worked out anyway. But on the other hand, once BIAB has crunched out the accompaniment I don't think simply playing it uses much CPU. It's the QT that is eating CPU as far as I know.

@ Comment 107 by goo...@ueleu.de,

Looks like great minds think alike....!!

Comment deleted

@kwg...@yahoo.com

Ooops, I think You got it first! I didn't read the latest comments as I changed the settings.

As I can say for now, no problems since I changed the source clock to Soundflower. I ran various audio apps at different channels, iTunes plays at channel 1+2 continously. Not only one dropout or distortion.

I forgot to mention: I own a MOTU UltraLite (first version) and use the lastest drivers.

I tried this yesterday and it ran fine for 4 hours. Heard a few crackles once, but they could have been some other piece of gear - not sure. The MOTU Ultralite I use at work has a lot of unshielded, in development audio gear connected to it. I used it to send some audio between MaxMSP and Logic over a soundflower/MOTU UL aggregate device for about 15 minutes and it worked. I left it alone and at the end of 4 hours i listened to a sine wave going from MaxMSP to logic and it was fine!

. . Glad to hear that the Clock Source toggle seems to be working for some of you. I do occasionally THINK that I hear a pop here or there. Nothing serious with that though. BUT...

For me, a couple of interesting things have popped up. You know how sometimes when you power on your gear and something will fail to be recognized by the mac? This has something to do with the order of how things power up and do their handshaking or whatever. I have not thought that much about it. If I have a problem with that I just restart or turn stuff off and back on and it seems to do the trick...

1) But since I have gotten my stuff going in the last few days, EVERY SINGLE TIME that I power up and get things rolling - NO SOUND. I have to go into Sound Prefs and re set the input/output back to MOTU, then I re-do the Audio Midi setup back to MOTU with INTERNAL Clock Source selected and then just get sound on a random QT video WITHOUT Soundflower. THEN I toggle the whole damn thing back to the Soundflower setup in Sound Prefs and then with the Audio Midi device window w/MOTU set to Soundflower for the Clock Source and then somehow, someway the thing starts working with my Soundflower/Quicktime setup and I get sound....wtf?

2) Another odd thing is that when I check my youtibe videos with the Soundflower setup now, the video screen pops up for a second and then goes completely WHITE as the sound plays. But if I re-set my sound back to the original NON-Soundflower setup and play the video then the screen comes back...again: wtf?

Any suggestions would be of great help.

Cheers.

@ arvi

did you set the buffer size in Sounflower to 2048?

no - 128 like everything else. 2048 would be ridiculous latency and defeat the purpose of using soundflower for me :)

I'm unable to use a buffer below 256 without instant jitter. At 256 and above I get drop out and distortion after only a few minutes.

Running a six core 3.33gHz mac pro, 16gb ram, RME FF400. Trying to monitor Logic Audio and route it to SpectraFoo. All software and drivers are up to date. Dont think I can do the slave clock workaround.

Hulp!

I use Soundflower to route my computer's audio through Ableton Live and out my soundcard. This way I can use my room correction software if I'm using my monitors, or TB Isone with my headphones. I like this setup as it normalizes the way I listen to audio and has really helped me advance my mixing and mastering skills. Alas I too suffer from the buffer problems.

I know how to set up an aggregate device. But I'm unsure of how to replicate my current Soundflower setup using it.

MacOS Audio Output (Soundflower 2ch) --> Ableton Input (Soundflower 2ch) --> [Processing through Ableton] --> Ableton Output (Presonus FireStudio Mobile) --> Out My speakers or Headphones.

Does anyone know why Cycling abandoned Soundflower? I think it's their most used/top product. I own Max for Live but I couldn't live without Soundflower.

You can't replicate that setup with aggregate devices and no soundflower.

I'd suggest you try JackOSX ( jackosx.com ) because it doesn't look like soundflower is ever going to be fixed :( If you're on 10.7 let me know and I'll send you a link to the latest JackOSX beta (or join their mailing list). While it takes more effort to set up, jack is more flexible and has a more robust clocking paradigm than soundflower (ie all software connected to jack get their audio clock from jack, which gets it from whichever audio hardware you slave it to) rather than the soundflower paradigm where different softwares can get audio clock from various different sources and soundflower or the OS tries to reconcile them. JackOSX does create some CPU overhead though.

I have the same exact trouble when trying to route my Maschine audio to Reason. After a determinate and fixed amount of time, the audio "slowly" disappears through crackling. This is such a shame since I was loving soundflower for this setup.

Just to add some info about my setup, I created an aggregated device with an Apogee Duet 2 USB and (sometimes) an Alesis IO 14. Changing the clock source doesn't solve the issue.

Echoing Comment 118, to me this definitely behaves like a clocking problem in which some parts of the system are clocked by one source and some by another. The clocks are nominally at the same frequency but one is not truly slaved to the other, and as a result they slowly drift: something tries to keep them locked together but eventually it fails and drops a cycle over a period of several seconds. AFAIK, the situation is not unique to SoundFlower, but in other cases you can usually solve it by careful analysis of clocking sources throughout the system.

@Elrik

could you please explain better what do you mean? I'm really interested in finding a solution. Does that involve the midi clock sync between two DAWs? or however, how/where could you set this clocks aside from Audio Midi setup?

Thanks

no - not MIDI sync clock - that is too coarse probably, but actually digital audio click at eg 44100 cycles/second. The inherent problem is that soundflower seems to generate its own clock and not get it from a physical audio device. if soundflower just had a setting to get its clock from somewhere, like Jack OS X does, it might work. I don't know who designed it thinking that it would nor need a clock input option. in any digital audio system there must be one master clock only. otherwise you're asking for trouble or a lot of effort in conforming one clock to another that may or may not always work.

The reason the trick with the MOTU hardware works is that MOTU's drivers allow you to select any audio interface connected to the system as the clock source for their hardware, something that few consumer or pro-sumer audio interfaces allow. If you select soundflower, then it works because the MOTU hardware slaves to soundflower.

Digital audio reall needs clocking, because its just a bitstream - in a simplified case, if you get off by one bit as to where one piece of equipment or software thinks your samples start and end vs another piece of gear/software, then everything goes to hell in a handbasket. Worst case suddenly it could be that all samples lose some precision and some are randomly very loud, as the least significant bit becomes most significant… basically you need one clock otherwise they drift. And drift compensation, while possible, is nowhere near as robust in my experience.

I'm not 100% positive, but after some test, it seems that setting in the aggregate device the clock to Soundflower and resample in the duet create a stable environment.

However I must say while I had the problem syncing two audio daws (Maschine and Reason) I rarely exceed the 5-6 minutes recording duration.

@Elrik, @ arvidtp does that make sense to you?

to add some extra info, Duet USB drivers doesn't allow to sync to an external clock, so if it works it is thanks to the resampling in the aggregate device, I guess

after long tryouts I can say the most stable configuration is Apogee Duet master clock, soundflower resample (drift correction). But sometimes I need to reset sample frequency, unfortunately

I had terrible problems using Soundflowerbed to control volume via DisplayPort audio device in my Dell U2713HM, with the sound going to distorted after short time due to clock drift.

I came up with this solution for now: Create an aggregate device in audio midi setup containing soundflower and your real audio device. Then set main device to soundflower and enable drift correction on the real device. Now choose the aggregate device as the output device in Soundflowerbed.

So far I'm running stable with a buffer size of 128.

I think this works, because now OS X correct the clock drift between soundflower and the actual audio device.

Oh forgot to add, that I set both channel on the soundflower device to roughly 75% volume, otherwise the signal distorts, not sure why.

The cause for the echo/feedback is commit 274948728ff7dbe164d8a53b5043d719e6fbcbf6.

Here is why:

The "clone" feature cannot be disabled due to a number of bugs in the coding as well as conceptual errors in the cloning as such. Because of that, the signal from the soundflower conduit is attenuated according to the volume control, then fed into the aggregate device. The attenuated signal is cloned into all channels of this device, which includes the soundflower input. As the feedback portion has already been attenuated, feedback diminishes rapidly as a function of volume and block size.

Near 100% volume, attenuation it reduced to the point where it becomes clearly audible and at 100% unchecked feedback results.

Reverting the above commit made the audio map (it was rendered unusable by this commit) work so the signal can be routed to the physical card only, while being properly resampled.

That commit needs to be backed out and redone properly.

As an additional item of information, output devices having more channels than the input are not properly zeroed out. This causes random bytes in the buffer which become subject to the same feedback loop, causing noise when switching to the aggregate device, which lasts for a few seconds and then diminishes through the above illustrated feedback mechanism.

I submitted a pull request for the fix to this issue.

I searched for hours trying to find a solution to this issue. A friend suggested an idea that he had been shown. Setup soundflower as a midi simultaneously with your regular output. Therefore instead of routing through soundflower for your output (which seems to cause the issue), it routes each as their own output. So far seems to be working on my recording meters without the overdrive static. Here is the how to on this.

http://apple.stackexchange.com/questions/50904/if-we-use-soundflower-to-record-the-systems-audio-output-then-we-cant-hear-it

If you also are looking for more control I found software from Prosoft that allows for audio adjustment using hear or individual app control through hear or soundbunny.

http://www.prosofteng.com/products/

Hope this helps everyone!

~Epoch Dream Productions

Just tried the aggregate device fix. It's working for me so far. Haven't had any issues for 15min or so.

fingers crossed

I'm running the Audio 4 DJ.

If this doesn't work, I guess I'll run my audio out of the front 1/4" channel B headphones out. At least it has a physical volume knob.

^^^ Update:

Well, this definitely did not work (I even set the clock to soundflower and made sure all sample rates were the same). After about 5 mins now, the sound will start crackling then fully crack down into distorted audio.

Too bad this app is not actively developed anymore, I haven't found any other that does what soundflower does. I'm using it to control the volume of my monitor, which has HDMI input but no volume control. The solution from Felix in post #127 works for me as well, but I also have to lower the output or input of soundflower2ch, otherwise I get the wierd echo/feedback. I guess better than nothing, cheers!

http://www.jackosx.com can be an alternative way.

Hi Julien, I spent a while trying to achieve this with jackosx, but it doesn't give me volume control through the keys. Would you care to explain how to achieve this? Thank you...

Hi Julien, I spent a while trying to achieve this with jackosx, but it doesn't give me volume control through the keys. Would you care to explain how to achieve this? Thank you...

I stumbled across this after spending a few years just not using this soundcard. I went through extensive troubleshooting on this issue and worked with Native Instruments. They identify the problem and stated "it may be fixed in a future update" years ago. This problem is not specific to Soundflower. The Native Instruments DACs are incompatible with any other external or virtual soundcards on any platform (ex. WIN, MAC, Linux, ASIO, Jack, etc). From what I can tell it has to do with some kind of drift compensation for bitrate. Creating an aggregate device of the NI device and your virtual/external card and enabling drift compensation does not fix it. Again, native instruments knows of the problem, and has chosen to do nothing about it.

Confirming issue with 10.8.5 / SF 1.6.6 / NI Kore 1 and NI Audio 4, not confirmed with internal MacbookPro2,1 soundcard though.

I have this same issue. It presents itself like this:

Macbook Pro ~2012 OSX Lion (Not Mountain Lion) Soundcard is external Mbox Pro 2, but this also happens on the built in card

Routing Reaktor into Reason via Soundflowerbed

Issue that occurs: trying to record out a 45 second song, the clicks and pops occur randomly but throughout.

I have not found ANYTHING that helps the clicks, be it buffer sizes, sample rates, or restarting. very sad, because I know of no other way to use my Native Instruments within Reason and this is simply unusable.

Have you tried JackOSX (http://www.macupdate.com/app/mac/16162/jackosx) ? Can you reproduce the problem with it ?