Audio levels automation in real time?

Hi,
Is it possible to automate the audio levels in REAL TIME? (using the mouse or an external mixer controller)
(automating one audio level at a time would be enough)
Thanks!

PPlease use feedback to make a feature request. We really do need some audio mixing tools, even simple ones.

Similar Messages

Can't perform Mute automation in real time

With touch or Latch mode on and pressing alternatively Mute on and off nothing is recorded here. What I miss?

That is how Touch mode works: it changes the parameter only when you touch (and hold) it. As soon as you'll let go, the parameter will return to the existing automation, or initial state. The 'release time' of touch mode can be set in Logic Pro>Preferences>Automation: Ramp Time parameter
Touch: Plays back automation in the same way as Read mode. You can modify the value of the chosen automation parameter by moving controls in Touch mode. After the fader or knob is released, the parameter follows existing automation on the track.
http://help.apple.com/logicpro/mac/10/#lgcpb1a6ab26

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

How to write data from query into Real time cube?

Hi All,
Can anyone explain me step by step how to write data into a real time cube from front end queries.
Thanks in advance

Hi
You can do this using Integrated Planning
You need to create a aggregation level on the Real Time infocube and can create Planning function/sequence, Variables if needed.
Then you can create query on this aggregation level and you can make the keyfigures Input ready in property pane and you can change the data and save it into cube.
Please find below help link which clearly explains step by step about Integrated Planning like creating input ready queries etc.,
http://help.sap.com/saphelp_nw70ehp1/helpdata/en/43/0c033316cd2bc4e10000000a114cbd/frameset.htm
Regards
Ravi

Does JMF support RTP packets being sent "Faster than real time"?

I have a situation where some stored audio is passed to a speech recogniser using RTP. This is all working well with JMF. However, since this operation is "offline" (i.e. no live person is actually speaking or hearing this audio stream) and the recognizer is capable of processing the audio very quickly, then the RTP stream could be sending the audio in "faster than real time". What settings in the following components would allow this?
DataSource _dataSource = Manager.createDataSource(source);
Processor _processor = Manager.createProcessor(_dataSource);
TrackControl[] trackControls = _processor.getTrackControls();
Codec codec[] = new Codec[3];
codec[0] = new com.ibm.media.codec.audio.rc.RCModule();
codec[1] = new com.ibm.media.codec.audio.ulaw.JavaEncoder();
codec[2] = new com.sun.media.codec.audio.ulaw.Packetizer();
((com.sun.media.codec.audio.ulaw.Packetizer) codec[2]).setPacketSize(160);
_processor.realize();
DataSource dataOutput = _processor.getDataOutput();
SendStream _sendStream = _rtpManager.createSendStream(dataOutput, 0);
_sendStream.start();
_processor.start();I tried "setRate" on the processor but this had no effect. getRate showed that it was still 1.0
Best Regards,
Jamie

I wrote my own RTP client in about an hour - (seemed simpler than navigating JMF options). It is very basic, but works as I want. The RTP server (the speech recognizer it able to consume the stream and gives exactly the same results).
package com.sss.mrcp;
import java.io.InputStream;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.util.Random;
public class RTP extends Thread {
     InputStream is;
     String address;
     int port;
     int localPort;
     public RTP(InputStream is, int localPort, String address, int port) {
          this.is = is;
          this.address = address;
          this.port = port;
          this.localPort = localPort;
     public void run() {
          try {
          DatagramSocket socket = new DatagramSocket(localPort);
          Random r = new Random();
          int sequenceNumber = r.nextInt();
          int syncId = r.nextInt();
          int timeStamp = 0;
          int len = 256;
          byte[] buf = new byte[len];
          int code = 0;
          int headerLength = 12;
          while ((code = is.read(buf, headerLength, len - headerLength)) > -1) {
               int i = 0;
               buf[i++] = (byte) 0x80; // version info
               buf[i++] = (byte) 0x08;     // 8=alaw,0=ulaw
               sequenceNumber++;
               buf[i++] = (byte) (sequenceNumber / 0x100);
               buf[i++] = (byte) (sequenceNumber % 0x100);
               timeStamp += (len - 12);
               int timeStampTop = (timeStamp / 0x10000);
               buf[i++] = (byte) (timeStampTop / 0x100);
               buf[i++] = (byte) (timeStampTop % 0x100);
               int timeStampBottom = (timeStamp % 0x10000);
               buf[i++] = (byte) (timeStampBottom / 0x100);
               buf[i++] = (byte) (timeStampBottom % 0x100);
               int syncIdTop = (syncId / 0x10000);
               buf[i++] = (byte) (syncIdTop / 0x100);
               buf[i++] = (byte) (syncIdTop % 0x100);
               int syncIdBottom = (syncId % 0x10000);
               buf[i++] = (byte) (syncIdBottom / 0x100);
               buf[i++] = (byte) (syncIdBottom % 0x100);
               DatagramPacket packet = new DatagramPacket(buf, code+headerLength, InetAddress.getByName(address), port);
               socket.send(packet);
               Thread.sleep(1); // this sets the speed of delivery "faster than real time"
          } catch (Exception e) {
               throw new RuntimeException(e);
}

Record Real Time output from a Audio Instrument to Audio Track

Hey,
Is there anyway to record in real-time the output of an Audio Instr?
I use a plugin that doesn't support automation, so I need to record the changes as they appear?
Any other solution?
D

Sound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower Sound FlowerSound Flower

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]$

How to Create an Epic Rap Battle of History in After Effects (Compositing, VFX, Green Screen, Audio), or Fix Audio, Speed, and Real Time Problems

Hello. As a film project, I am creating an "Epic Rap Battle of History", a parody video of a popular series on YouTube. In my film/video, I would like to use compositing so that there will be two of one character in one shot. I have used Premiere Pro CC for the main character sequences, but when I imported this project in After Effects, there was no audio at all from the song, and the canvas displaying the composition was moving very slowly, at about 50% of the video's actual speed. Also, in the top right corner, it says, "Not Real Time", and displays something like 7.39/29.96. I have looked at different web pages, and nothing seems to work.
I would appreciate it if someone could explain how to fix this all up for me, or especially if someone who has had experience creating an Epic Rap Battle could actually list a step-by-step process on how to create one.
If this helps, I am a beginner to Adobe, and work on a iMac tower. Thank you for your feedback.

Start here to learn After Effects: http://adobe.ly/AE_basics

Caltyp24_Call Type Real Time All Fields Service Level Today discrepancy

Hi All,
During our initial phase of deployment to CUIC reporting tool we want to offer similar features to our team to ensure smoother transition. However, I'm seeing discrepancy in data pulled by two different reporting engines - Webview 7.5 and CUIC 8.x. I have attached the screen shot of both highlighted in red.
Details:
CUIC/WebView Report: caltyp24_Call Type Real Time All Fields;
Version of CUIC: 8.5(4) build 1 (8_5_4_10000_99);
ICM Version: Release 7.5.1.0, Build 23684;
Column with Discrepancy: Service level today.
Any help much appreciated.
Thanks in advance,
Ramchand

Thanks very much for responding so quickly Kartik. The data is pulled from the ServiceLevelToday column in Call_Type_Real_Time table. Here is the query; I have truncated some columns for brevity:
SELECT Call_Type.EnterpriseName,
Call_Type_Real_Time.CallTypeID,
Call_Type_Real_Time.ServiceLevelCallsToday,
Call_Type_Real_Time.ServiceLevelToday,
Call_Type_Real_Time.TalkTimeToday,
Call_Type_Real_Time.ReturnReleaseHalf,
RouterCallsAbandToAgentTo5 = Call_Type_Real_Time.RouterCallsAbandToAgentTo5,
RouterCallsAbandToAgentHalf = Call_Type_Real_Time.RouterCallsAbandToAgentHalf,
CallsAtVRUNow = Call_Type_Real_Time.CallsAtVRUNow,
CallsAtAgentNow = Call_Type_Real_Time.CallsAtAgentNow
FROM Call_Type, Call_Type_Real_Time
where Call_Type.CallTypeID = Call_Type_Real_Time.CallTypeID
Order BY Call_Type.EnterpriseName, Call_Type_Real_Time.DateTime
We used the templates through CDN that mimic the real time and historic reports from Web View. This is the only real time report where we noticed the discrepancy in service level today column. We have the Standard CUIC license so I am unable to verify the report definition through CUIC. However, I should assume that both reports are referring to the same columns and tables. Do we have any alternatives? Please suggest…thanks

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.

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!

Export audio function in logic..can it be done in real time?

hey,
does anyone know if the export audio option can be done in real time as oppose to an offline bounce. there is a sonic difference between exporting files offline and actually bouncing them individually. Enough to actually make me consider going back to the old school way of bouning each track one by one. i obviously don't wnat to do that but and very dissapointed the hear the sonic difference between the two. If i do decide to do that wouald lla to the virtual inst be sample accurate? I know the exs 24 would be but what about spectrasonics stuff or other 3rd party virtual inst. does logic ahve and interanl engine that mkaes them sample accurate?
thanks in advance,
ej

hey justin
i'm not sure i i get what you mean. if you are saying that i could lets say take 16 outs of logic and digitally record them in the pt then that is true...however logic drifts when locked to timecode which becomes obvious when you have to do mulitple passes (i guess all daws do to so extent). also let say that if i did a stereo track bounce of the music and then imported that bounce into pt for the purposes of recording vox. if i then decide to import the tracks via analog or digital, the track will be out of time with the original trak and require me to start shifting vox to compensate...not fun.
the thing that i like about exporting and bounce is that i don't have to worry about timing issues that way..it's just that bouncing sounds way better that exporting.
if i could export in real time then that would solve my problem. I guess i now see why digi has ignored requests foe offline bouncing,
any thoughts would be appreciated
ej

Real-time mic level

Hi,
Not sure if this is the right subforum.
I'm trying to access the level of the mic through Java.
I don't need to record anything, I just want to know a relative scale of sound level.
Is this possible in real-time?

timvdalen wrote:
I'm trying to access the level of the mic through Java.
I don't need to record anything, I just want to know a relative scale of sound level.http://www.jsresources.org/examples/SimpleAudioRecorder.html
Looking at that code, you can rip out the recording part and just use the targetDataLine to determine the level by doing targetDataLine.getLevel

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]

Recording with audio playback slowed down 65% what speed do I need to speed the video back up to to get it into real time?

Ive recorded a test video with audio/music playback slowed down 65% what speed do I need to speed the video back up to to get it into "real" time?
If I'd slowed the audio playback down 50% I would make the video 100% faster.
If I'd slowed the audio playback down 100% I would make the video 200% faster.
So its obviously somewhere inbetween speeding up by 100-200%
Sorry but its melting my brain today and I cant remember the calculation.
thanks

hey all
Thanks for the advise on this. I will explain a little more what am have/am intending to do.
I have used Audacity to slow the song down (Effect>Change Tempo)
Audacity will allow me slow the original song down by up to 99% (Or anything in-between up to 3 decimal places)
Unfortunately I am working to a time restraint of 12 minutes per clip (as my DSLR will only allow up to a 12 minute clip).
The song at normal speed is 4 minutes long.
Originally I slowed the audio down by 50% (Making the audio 8 minutes long), lip synced to it using a rasta blaster and recorded the video using my DSLR at 720p 50fps.
I then imported the video and audio I had been lip syncing to into FCPX.
I sped up the video by x2 (equivalent to speeding it up 200%)
Retime>Fast 2X
Layered the PROPER normal speed audio back over the top and all was fine.
Test video complete.
(You can ignore the 65% slowed down audio I mentioned earlier if you like, that was just an example of a slower than 50% audio test)
Firstly what I really want to do is to be able to slow the AUDIO down as much as possible (But still within the restraints of only being able to adjust FCPX retiming within a whole 1 percentage in the final video edit), but have the AUDIO of the song as near to/under 12 minutes as possible.
I will then re-record the video again to this new slowed down audio.
Secondly I will speed up the video from 12 minutes to 4 minutes.
Thirdly I will layer the correct 100% Normal AUDIO on top.
The FINISHED video should then look like everyone is wizzing around at high speed but the guitar playing/lyrics look like they are being played/sung at the normal speed.
Whatever happens (due to the constraints of FCPX) I need the calculations to work so that FCPX is retimed exactly to a whole 1% for the final video to sync up correctly.
So what I need to know in a nut shell is:-
What calculation do I use to work out the percentage to slow a 4 minute song down to a 12 minute song?
What calculation do I use to work out the percentage of how much I need to then speed the 12 minute video up to turn it back into a 4 minute video?
My mind is melting

Audio levels automation in real time?

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