Possible to make a real-time audio analysis program?

Hi, I was wondering if it is possible to use flash to create a swf file that detects audio levels (or certain frequencies) so that on this event it could trigger a movie clip, or timeline event, if not, does anyone know what could be used to do this?
Thanks!

The tree vi for the above are in these .llb. Thanks
Attachments:
6430 project unedited.llb ‏1267 KB
6487 project.llb ‏221 KB
3390 project.llb ‏2356 KB

Similar Messages

Real-Time Audio Spectrum in FCPro?

Is it possible to do a real-time audio spectrum analysis (i.e., some sort of audio frequency histogram) in Final Cut Pro 4.5? I have some interview audio that has some high-frequency noise, and I'd like to easily be able to tell where in the spectrum it's happening.
Thanks.

True, but the process of exporting the appropriate audio clip to another app, finding the offending portion, then coming back into FCPro to filter it out seems like a lot of work considering that FCPro does have a lot of other audio functionality--and a video spectrometer--built-in. It would be really great if FCPro did this!

Sndpeek. real-time audio visualization

Homepage:
http://soundlab.cs.princeton.edu/software/sndpeek/
Screens:
http://soundlab.cs.princeton.edu/images/sndpeek.jpg
http://soundlab.cs.princeton.edu/images/sndpeek6.jpg
http://soundlab.cs.princeton.edu/images/sndpeek2.jpg
http://soundlab.cs.princeton.edu/images/sndpeek3.jpg
sndpeek is just what it sounds (and looks) like:
* real-time 3D animated display/playback
* can use mic-input or wav/aiff/snd/raw/mat file (with playback)
* time-domain waveform
* FFT magnitude spectrum
* 3D waterfall plot
* lissajous! (interchannel correlation)
* rotatable and scalable display
* freeze frame! (for didactic purposes)
* real-time spectral feature extraction (centroid, rms, flux, rolloff)
* available on MacOS X, Linux, and Windows under GPL
* part of the sndtools distribution.
i have the intention to make the PKGBUILD, but it has no ./configure and its default is to install on /usr/local/
# Contributor: Your Name <[email protected]>
pkgname=sndpeek
pkgver=1.3
pkgrel=1
pkgdesc="real-time audio visualization "
url="http://soundlab.cs.princeton.edu/software/sndpeek/"
arch=('i686' 'x86_64')
license=('GPL')
depends=(libsndfile)
provides=(sndpeek)
source=(http://soundlab.cs.princeton.edu/software/sndpeek/files/$pkgname-$pkgver.tgz)
md5sums=('0ad03fa135bf819fb5971fde015526b4')
build() {
cd $srcdir/$pkgname-$pkgver
./configure --prefix=/usr
make || return 1
make DESTDIR=$pkgdir install || return 1

well changed the PKGBUILD a little... but still got errors...
# Contributor: Leandro Chescotta <[email protected]>
pkgname=sndpeek
pkgver=1.3
pkgrel=1
pkgdesc="real-time audio visualization"
arch=('i686' 'x86_64')
url="http://soundlab.cs.princeton.edu/software/sndpeek/"
license=('GPL')
depends=('libsndfile')
source=(http://soundlab.cs.princeton.edu/software/sndpeek/files/$pkgname-$pkgver.tgz)
md5sums=('0ad03fa135bf819fb5971fde015526b4')
build() {
cd $srcdir/$pkgname-$pkgver/src/sndpeek
./configure --prefix=/usr
make linux-alsa || return 1
make DESTDIR=$pkgdir install || return 1
output:
[aleyscha@aleyscha 51 sndpeek 22:15]$ makepkg -f
==> Making package: sndpeek 1.3-1 i686 (Fri May 15 22:15:45 ART 2009)
==> Checking Runtime Dependencies...
==> Checking Buildtime Dependencies...
==> Retrieving Sources...
-> Found sndpeek-1.3.tgz in build dir
==> Validating source files with md5sums...
sndpeek-1.3.tgz ... Passed
==> Extracting Sources...
-> bsdtar -x -f sndpeek-1.3.tgz
==> Removing existing pkg/ directory...
==> Entering fakeroot environment...
==> Starting build()...
PKGBUILD: line 16: ./configure: No such file or directory
make -f makefile.alsa
make[1]: Entering directory `/home/aleyscha/bin/arch_packages/sndpeek/src/sndpeek-1.3/src/sndpeek'
gcc -D__LINUX_ALSA__ -D__LITTLE_ENDIAN__ -I../marsyas/ -O3 -c chuck_fft.c
gcc -D__LINUX_ALSA__ -D__LITTLE_ENDIAN__ -I../marsyas/ -O3 -c RtAudio.cpp
RtAudio.cpp: In member function 'void RtApi::openStream(int, int, int, int, RtAudioFormat, int, int*, int)':
RtAudio.cpp:234: error: 'sprintf' was not declared in this scope
RtAudio.cpp:239: error: 'sprintf' was not declared in this scope
RtAudio.cpp:244: error: 'sprintf' was not declared in this scope
RtAudio.cpp:250: error: 'sprintf' was not declared in this scope
RtAudio.cpp:257: error: 'sprintf' was not declared in this scope
RtAudio.cpp:339: error: 'sprintf' was not declared in this scope
RtAudio.cpp:341: error: 'sprintf' was not declared in this scope
RtAudio.cpp: In member function 'RtAudioDeviceInfo RtApi::getDeviceInfo(int)':
RtAudio.cpp:355: error: 'sprintf' was not declared in this scope
make[1]: *** [RtAudio.o] Error 1
make[1]: Leaving directory `/home/aleyscha/bin/arch_packages/sndpeek/src/sndpeek-1.3/src/sndpeek'
make: [linux-alsa] Error 2 (ignored)
cp /usr/local/bin/; chmod 755 /usr/local/bin/
cp: missing destination file operand after `/usr/local/bin/'
Try `cp --help' for more information.
==> Tidying install...
-> Compressing man pages...
-> Stripping debugging symbols from binaries and libraries...
==> Creating package...
-> Generating .PKGINFO file...
-> Compressing package...
==> Leaving fakeroot environment.
==> Finished making: sndpeek 1.3-1 i686 (Fri May 15 22:15:50 ART 2009)
Press any key to continue...
[aleyscha@aleyscha 52 sndpeek 22:15]$

Adobe Director 11.5 - Real-Time Audio Manipulation?

Hi all,
Firstly, I'm enrolled in a Multimedia Design course in College, and our College is using Director 11.5, hence why I need advice specifically on the older version, not the current one.
Our assignment for one of the courses is to author an interactive visual dictionary - using Director 11.5 - on a topic of our choice. In my group's case, it was the workings of a soundboard.
In an attempt to really increase the level of interactivity and make a superb addition to future portfolios - as well as showing off - we want to include an interactive 'soundboard simulation' which users can play with. There will be three separate audio inputs that play continuously, and 5 dials to increase or decrease certain effects. Those dials/effects are:
EQ - Hi (High Pass Filter)
EQ - Lo (Low Pass Filter)
FX - (Reverb or Echo)
Pan - (Left to Right)
Level - (Volume)
The idea is that as the user adjusts the dials, the audio is effected in real-time with a multitude of possible effects being applied. Is this remotely possible with Director? As far as coding in Director, I'm still a relative novice, but I have a good head for code and a lot of the principles are the same as in Javascript (which I'm also learning and I must say, pretty good at), it's just the syntax itself which may prove troublesome to wrap my head around at times.
But for now I'm just asking if this is even possible. At our current level of Director proficiency, we're sort of shooting for the moon, but I don't see anything wrong with that!
The only page I've found listing various audio filters and parameters in Director is this: http://help.adobe.com/en_US/Director/11.5/UsingScripting/WSCB4744EA-6E6E-4b85-A946-E2C6E3A 4450D.html
However, I'm not certain that those can effect audio real-time (even just an 'onMouseUp' effect would be fine, but we'd much prefer it be truly real-time).
Any advice, hints, tips, tricks, or directions would be very much appreciated!

Hello Peregrine.2976,
Here is a link to a .dir file that adjusts volume in real time
by setting a variable and calculating volume in play frame as user
slides volume button. Can show more about sound control if needed.
http://www.theclickpoint.com/volume/volume.dir
Peace,
Jack

Real-time audio chat

Hello folks!
it's my first post, so, please, forgive me if I'm breaking some rule..
I've thought about an web application I want to develop, but I'm not sure if it's possible and that's why I'm posting my question on this forum.
I want to develop a chat which allows a real-time conversation by voice using JMF (or another solution).. but I don't know how to get started. (is it possible without applets?)
if you could send me some links to tutorials or any sort of aid, it would be really nice.
Thanks in advance!!
p.s. I know my english isn't so good.. so forgive me again
greetings from Brazil!!
Edited by: luis.celestino on May 20, 2009 5:24 AM

luis.celestino wrote:
I want to develop a chat which allows a real-time conversation by voice using JMF (or another solution).. but I don't know how to get started. (is it possible without applets?)If you're wanting to deploy an application like this on the web, you could certainly do it using JMF, and if you're not wanting to do an applet, you could do it as a Java web-start application.
Here are some JMF resources to get you started learning how to do stuff. Pay close attention to the ones talking about sending audio/video data via RTP, and receiving it.
API
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/apidocs/]
Examples
[http://www.cs.odu.edu/~cs778/spring04/lectures/jmfsolutions/examplesindex.html]

Is it possible to make a real 3D path in Motion?

Hi guys,
it's still me...
Now I'm trying to make a real 3d path using the paint stroke.
The problem is the paint points have only 2 parameters (X and Y) and I can paint it only on a 2d plain and not in a 3d world.
I would make something similar to:
http://www.youtube.com/watch?v=GBKAk53tMNM
Is it possible using Motion?
Thanks in advance,
Alessandro

Paint strokes are always on a 2D plane in 3D space, and always in "local 3D" which means they cannon interact with other objects (unless you keep them as 2D).
However, you can animate particle emitters and replicators along a motion path in full 3D space and wrap around objects.

After Effects Playback 27 fps (Real Time) Audio is Slow

Hello,
I have Adobe After Effects CS 5 running on an HP Z600 8 core workstation (12 GB RAM) on Windows 7.
When I import any video AVI, MOV, etc and I preview with RAM the audio plays slow and it reports fps: 27 (real time)
When I link comps from premier I think it is causing lip sync issues.
My co-worker has the identical computer and when importing the same video, his reports fps: 29.97 (real time) and audio play fine.
I have looked around for settings that may affect this but nothing seems to help.
Multi-threading? Open GL issue?
Any Ideas?
Thanks,
Greg

> Sorry, I just figured it out.
If you think that what you just figured out might be useful to others, feel free to elaborate here. If not, then I'll go ahead and remove this thread if you don't mind.

Is it possible to download the Real Time Collaboration ActiveX control?

We would like to deploy the Real Time Collaboration ActiveX control to our user's PCs using SMS. Our users do not have Admin rights on their PC, so the ActiveX control will not install automatically. I check on the SAP site for the download, but could not find it. Is the ActiveX control available for download somewhere?
Thanks

Hi,
Read this link:
http://help.sap.com/saphelp_nw70/helpdata/EN/24/ae5e77711b431c887aa5193af8d851/frameset.htm
http://help.sap.com/saphelp_nw04/helpdata/EN/45/cc3541afe5db2be10000000a1550b0/content.htm
You can also verify the download as follows:
       1.      In the Web browser of the user, from the Tools menu, select Internet Options.
       2.      In the General tab, click the Settings button.
       3.      In the Settings window, click the View Objects button.
       4.      In the Download Program Files window, if any of the following appear in the Program Files list, then the ActiveX control was downloaded:
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        ExpApplSelect
        ExpApplShareX
Patricio.

Real time audio streaming

Hi, I need to develop a j2me softphone, is there someone who can tell me how to do audio streaming ? I already read jsr135 but nothing important.
I would appreciate any help.

Hi Daniel,
I'm encouraged to receive a response!
I can't see how the advent of a new codec will affect the issues that I mention above. You still need an unreliable protocol to transport the Speex data! Datagram support is optional in both MIDP2.0 and it's my understanding that it is rarely implemented. The objection about Bluetooth remains - L2CAP support is optional, and setting the flush timeout on an L2CAP connection is impossible.
Then there is the question of playing and recording at the same time. The anecdotal evidence seems to be that even though you can use threading to start two players at the same time, as soon as you try record and play at the same time, an exception is thrown (sometimes the emulators work but after deploying on the device the exception is thrown).
Finally, you mention that there are a number of Java ME VOIP applications out there. Do you have links? Personally I can't find any. What I can find is a number of products that operate on different platforms:
eg Fring that has Symbian, iPhone OS, Windows Mobile and Java ME versions.
What is telling is that the Java ME version does not support voice, while the other versions do. The same applies to Nimbuzz and Talkonaut. Extensive googling has not revealed any exceptions to this rule.
If you can prove me wrong it'll make my day :).
Cheers,
Fritz

Real Time Audio Mixing.

Hi!
I would like to mix 2 rtp stream into one. Is there any possibility?
Thanks in advance!:D

Hi captfoss!
I got to create de MergedDataSource, but when I tried to create the sendStream
sendStream = RTPmanager.createSendStream(outputProcessor.getDataOutput(),0);          it throws a UnsupportedFormatException.I use the index 0, because I want to mix all the track in one stream.
The audio format used in the track is ALAW_RTP. So, in outputProcessor I convert the track into this format and I add this format to the RTPManager object.
To do convert all the track of DataSource object into ALAW_RTP format I do this:
try {
     outputProcessor = Manager.createProcessor(mergeDS);
} catch (IOException e) {
       gestor.notifyEvent(createCallErrorEvent(ErrorConstants.IO_ERROR,"IOException. Error en la creacion del Data Source o del outputProcessor al establecer la comunicación."));
} catch (NoProcessorException e) {
     gestor.notifyEvent(createCallErrorEvent(ErrorConstants.CREATE_PROCESSOR,"NoProcessorException. Error en la creacion del outputProcessor al establecer la comunicación."));
outputProcessor.addControllerListener(this);
try
     logger.debug("["+callID+"]OutputProcessor.configure()");
     outputProcessor.configure();
}catch(Exception e){
     logger.error("["+callID+"]establecerComunicacion - outputProcessor.configure"+e.getMessage());
}when the outputProcessor is configured:
if(audioFormat!=null)
     TrackControl track[]=outputProcessor.getTrackControls();
     for (int i = 0; i < track.length; i++) {
          track.setEnabled(true);
     Format [] alawRTPFormat = alawrtpPacketizer.getSupportedOutputFormats(null);
     if(audioFormat.matches(alawRTPFormat[0]))
          Codec[] codec = new Codec[2];
          codec[0] = alawEncoder;
          codec[1] = alawrtpPacketizer;
          for (int i = 0; i < track.length; i++) {
               track[i].setFormat(audioFormat);
               logger.debug("["+callID+"]outputprocessorConfigured - Habilitada pista : "+i);
          try {
               for (int i = 0; i < track.length; i++) {
                    track[0].setCodecChain(codec);
          } catch (UnsupportedPlugInException e) {
               gestor.notifyEvent(createCallErrorEvent(ErrorConstants.UNSUPPORTED_PLUGINGS,"UnsupportedPlugInException al modificar la cadena de codecs del outputProcessor."));
          } catch (NotConfiguredError e) {
               gestor.notifyEvent(createCallErrorEvent(ErrorConstants.NOT_CONFIGURED_PROCESSOR,"NotConfiguredError al modificar la cadena de codecs del inputProcessor."));
     else
          for (int i = 0; i < track.length; i++) {
          track[i].setEnabled(true);
try
     logger.debug("["+callID+"]outputProcessor.realize()");
     outputProcessor.realize();
} catch (Exception e) {
     logger.error("["+callID+"]outputProcessorConfigured - realize."+e.getMessage());
when the outputProcessor is realized:     outputDS = (DataSource) outputProcessor.getDataOutput();
     initSessionManager();
and initSessionManager method is this:private void initSessionManager()
     try {
          logger.debug("["+callID+"]initSessionManager - RTPmanager.createSendStream() ");
          sendStream = RTPmanager.createSendStream(outputProcessor.getDataOutput(),0);
     } catch (NullPointerException e) {
          e.printStackTrace();
     } catch (UnsupportedFormatException e) {
          gestor.notifyEvent(createCallErrorEvent(ErrorConstants.UNSUPPORTED_FORMAT,"UnsupportedFormatException. Error el formato no es soportado en la sesión."));
     } catch (IOException e) {
          gestor.notifyEvent(createCallErrorEvent(ErrorConstants.IO_ERROR,"IOException Error al crear el SendStream."));
     } catch (Exception e) {
          logger.error("["+callID+"]InitSessionManager - createSendStream. Exception:"+e.getMessage());
     try {
          logger.debug("["+callID+"]InitSessionManager - sendStream.start() ");
          sendStream.start();
          logger.debug("["+callID+"]outputProcessor.start()");
          outputProcessor.start();
} catch (NullPointerException e) {
          e.printStackTrace();
     }catch (IOException e) {
          gestor.notifyEvent(createCallErrorEvent(ErrorConstants.IO_ERROR,"IOException Error al comenzar a enviar el SendStream."));
     } catch (Exception e) {
          logger.error("["+callID+"]InitSessionManager - sendStream.start."+e.getMessage());
Any idea about it?
Thank you very much for your help!:D

How can I get real-time audio (think Bloom from Brian Eno out of my ipad and into garageband on my mbp?

I'm a visual artist and an audio novice who would like capture some spontaneous Bloom or Scape musical noodlings. Since neither Bloom nor Scape allow users to save to a universal audio format, I'm hoping that there's a way for me to get those noodlings out of my iPad and into GrageBand on my MBP
Thanks, in advance, for any help

See Recover your iTunes library from your iPod or iOS device.
While I agree that it would be nice if it was easier to transfer media from your Apple devices they should never be used as the sole location for any of your media. They are too easily lost, stolen, damaged, or corrupted. The user tip above also includes a link to a suggested library backup strategy so that you can avoid being in the same position in future.
tt2

T520i Sound problem (real-time audio)

I`m using my Lenovo T520i i7-2630QM with 8GB Ram for my DJ needs.But i have pops, hisses, crackles and pauses. I disabled the wireless adapter in order to get better results and it`s working but in every 15 min or 30min there is crackles and pops.I`ve run the LatencyMon http://imageshack.us/f/861/212324333434344.jpg/. the highest dpc routin execution is iaStor.sys. any ideas ?

Hi vlad5657
Try updating to the latest Intel Chipset driver using System Update.
You could also try downgrade to HP's Intel LAN driver. (High % causes from this)
http://h20000.www2.hp.com/bizsupport/TechSupport/SoftwareDescription.jsp?lang=de&cc=de&prodTypeId=32...
&swLang=18&mode=2&taskId=135&swItem=ob-96025-1
Related issues:
http://forums.lenovo.com/t5/T400-T500-and-newer-T-series/DPC-latency-audio-mouse-skip-and-stutter-on...
http://forums.lenovo.com/t5/W-Series-ThinkPad-Laptops/Workaround-for-high-DPC-issues-for-X-T-and-W-0...
You could find more on similar cases on this forums.
By the way, since you are a DJ, you could achieve better sound / customization through Dolby audio drivers instead of using default Conexant audio drivers.
Hopefully, 2012 would be a better year for all of you!
Peter
ThinkPad: W520 (4284-A99)
=============================================
Does someone’s post help you? Give them kudos as a reward, as they will do better to improve
Mark it as solved if the solution works for you, so it could be reference for others in the future
Dolby Home Theater v4 (Essential/IdeaPad/ThinkPad)
http://forums.lenovo.com/t5/General-Discussion/Dolby-Home-Theater-v4-for-most-Lenovo-Laptops/td-p/62...

How to make labview real time understand sequence context

Does anyone know how to make Labview RT to understand teststand sequence context? My sequence context is defined as "TS.SequenceContext" as the screenshot in the attachment.
If we need to point a VI for this Sequence Context then which VI from teststand to use? and do we need to add this VI to the Labview Projectr Explorer in RT System to build the project?
I'm using Labview 2009 and teststand 4.2.1.
Thank you very much
Solved!
Go to Solution.
Attachments:
LVRT_TS.SequenceContext error.doc ‏132 KB

LabVIEW RT does not support ActiveX which is required for TestStand.

Instalacion ni real time module 2013 programa de 32 bits y computador de 64 bits

buenas noches al intentar instalar este modulo con programa labview 2013 de 32 bits y computador de 64 bits no me deja instalar no se porque razon si se supone que todos los modulos funcionan para la version del programa labview 2013 de 32 bits por favor espero respuesta o ayudaaa

Saludos david1348,
Gracias por utilizar los Foros de Discusión de National Instruments, me gustaría revisar el Software de NI que se encuentra instalado en su equipo, ¿podría compartir con nosotros un reporte técnico de NI MAX?, el siguiente link explica cómo generar dicho reporte:
Using MAX Technical Report to Document Measurement & Automation Explorer (MAX) Configuration Information:
http://digital.ni.com/public.nsf/allkb/271F252B4EF0A2E0862570E70056A1E4
Quedo al pendiente, saludos cordiales.
David P.
National Instruments
Applications Engineer
www.ni.com/soporte

Parallel computation - real time analysis

Im dooing real time image analysis, however my analysis program is the bottleneck at the moment so i want to increase its speed. Since I need an increase of about 60-80% the only solution would be dooing the analysis in parallel on a dual core cpu (currently running on AMD64 3700+).
The most time consuming computations in my program are two convoluitons so my idear is to have them running in parallel. I want to grab one image, do the first convolution and then send the result to a shift register. Next the program should (in parallel) grab a new image and to the convolution - and do the second convolution on the first image.
Is this possible?
As you can see in the attached vi Im trying to transfer the images in shift registers is this the right way?
How do I do the wireing of the error correctly?
How do I make sure this can actually run in parallel?
Hope someone can help me.
Simon
PS: This version of the program does not grab images, it just loads in previosly saved images. That way I can test the results.
Attachments:
speed test4.vi ‏52 KB

Hi, if you want to have thoses convolute functions running in parallel, you will need to give them their own independent error in. That means the seconde convolute function shouldn't get its input from the first one.
I have modified your code so the two convolute functions are now running in parallel. From the attached result.jpg I can see a improvment at the VI execution time.
But I'm not sure about result, it seems that somthing is missing from the output array.
Please let me know if you have found a better solution to this problem.
Regards
Dennis Morini
Field Sales Engineer
National Instruments Denmark
http://www.ni.com/ask
Attachments:
speed test4_mod.vi ‏53 KB
Result.JPG ‏93 KB

Possible to make a real-time audio analysis program?

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