CVP wave file change

When replacing wav files in CVP 8, what service(s) needs to be restarted for the new wav files to play? We replaced an old wav file with a new one. Somehow the old wav file still plays. I'm guessing that it is cached and either the server needs to be rebooted or a service needs to be restarted.
Any comment is appreciated. Thanks!
Richard

Hi,
No need to restart any services from the CVP or from the media server. The prompts get cached in the VXML gateway. The default time set to refresh the prompts is 864000 seconds (10 days). When uploading the new prompts in the CVP you need to run the command "set http client cache stale" in the VXML Gateways, so it shall look out for any new prmpts in the media server
Thanks
Dass

Similar Messages

How do I input a wav. file, change the freq (cap the high freqs) and then save the new waveform as a new wav file again?

Using Labview 7

Hi Alexi,
I have found a vi in the examples and slightly modified it for your purposes. Basically this is just taking the idea from the post above and partially implementing it into a vi.
I have added a couple of notes to help you.
If you need any further help with this, then I will be watching this post so please feel free to post back with more questions.
Best regards,
Peter H
Applications Engineer
National Instruments
Attachments:
Wave_File_Player.vi ‏192 KB

How do I change the key of a wav file?

I'm using Apple Loops Utility and the wav file plays the correct key & tempo there. But when I save it, and open it up in LE or add it into the arrangement, it's playing the original key? The only thing that is correct is the tempo. Can you change the key and save the change using Apple Loops Utility?

I'm confused. Are you saying that in Apple Loops Utility the wav plays in the key and tempo of that wave, but when you add the same wav to LE it plays in a different key? You said in LE it plays in the "original" key. Seems like that is what you would expect. I would think it would play in the "original" key in both ALU and LE.

Basic tcl and wav files required in VG for a stanalone cvp architecture

hy all
I am very new to this field and still learning can someone help me gain knowledge on this subject my question is
What are the files needed in VG for supporting a standalone CVP architecture ?   (in specific the VXML,tcl and wav files ??
What are the use of the following services run in an ICM router ?
rtsvr, dbagent, testsync, router, ccagent, mdsproc
What are the use of the following services run in an ICM logger ?
configlogger, csfs, listlogger, recovery, replication

A stand-alone CVP VXML solution does not need ICM at all. You simply set up the services on the gateway
service CVPSelfService flash:CVPSelfServiceBootstrap.vxml
service helloworld flash:CVPSelfService.tcl
param CVPSelfService-port 7000
param CVPSelfService-app HelloWorld             <<<<<<<< whatever the VXML app is called
paramspace english location flash
param CVPPrimaryVXMLServer 164.103.128.20 <<<<<<<< primary VXML server
param CVPBackupVXMLServer 172.20.142.20   <<<<<<<< backup VXML server
I usually also install
service cvperror flash:cvperror.tcl
service ringtone flash:ringtone.tcl
as well.
Regards,
Geoff

Changed wav file names – files unrecoverable

Hi. I was asked to import some important imterviews made on an iPod mini and an ExtremeMac Micromemo onto a Mac, but, as requested, I change the file names first. Since changing the file names the files are completely unplayable, undownloadable and unrecoverable. Please, can anyone help.
Matthew Whiting

Hi Chris
What I did was to change the file names of the wav files in the Finder while they were still on the iPod Nano, before downloading them to iTunes. The iPod Nano uses a hybrid file system, and once the file names were changed iTunes refused to recognize the files. The Finder and Mac OS X flags them as wav files but cannot read them. All other software (players, audio apps and file recovery apps) that we have attempted to use, whether on Mac OS X, Windows XP and Vista, and Linux Red Hat has either crashed, failed to read the files or ignored them. Changing the file names back to an arbitrary time, similar to the original, still does not help.
It's hard to appreciate the problem, but two of us with many years of Mac (and other IT) experience spent three hours yesterday, on a total of 6 computers, trying to get something to read or recover these files, without any luck whatsoever.
Thanks for your interest.
I might be able to get my hands on some industrial strength recovery software, today, and then we'll see. The problem, however, is that the data of the files is intact, but absolutely nothing succeeds in reading them as wav files.
Matthew Whiting

Change wav file names on iPod - unreadable

Hi. I was asked to import some important imterviews made on an iPod mini and an ExtremeMac Micromemo onto a Mac, but, as requested, I change the file names first. Since changing the file names the files are completely unplayable, undownloadable and unrecoverable. Please, can anyone help.
Matthew Whiting

Hi Chris
What I did was to change the file names of the wav files in the Finder while they were still on the iPod Nano, before downloading them to iTunes. The iPod Nano uses a hybrid file system, and once the file names were changed iTunes refused to recognize the files. The Finder and Mac OS X flags them as wav files but cannot read them. All other software (players, audio apps and file recovery apps) that we have attempted to use, whether on Mac OS X, Windows XP and Vista, and Linux Red Hat has either crashed, failed to read the files or ignored them. Changing the file names back to an arbitrary time, similar to the original, still does not help.
It's hard to appreciate the problem, but two of us with many years of Mac (and other IT) experience spent three hours yesterday, on a total of 6 computers, trying to get something to read or recover these files, without any luck whatsoever.
Thanks for your interest.
I might be able to get my hands on some industrial strength recovery software, today, and then we'll see. The problem, however, is that the data of the files is intact, but absolutely nothing succeeds in reading them as wav files.
Matthew Whiting

AC - change incoming call alert wav files

Hi,
With CCM4.1(3)sr3a I need to change the audible alerts that AC plays when an incoming call arrives.
I've found the source files on each attendants PC (C:\Program Files\Cisco\Call Manager Attendant Console\audio), copied in a replacement & renamed it to Cisco's original file name.
However, on startup AC is copying the original files back from CCM.
If I changed the files on CCM then this would solve my problem but I can't find them.
How can I change the audio file source ?
Any help appreciated,
Paul.

The current capabilities and features of MoH include:
MoH multicast and unicast streaming service
Music streaming service for "user" hold and "network" hold
51 sources per media convergence server (MCS)
Fifty continuously looping .wav file sources
One real-time streaming source
Each source configurable as either unicast or multi-cast stream
Support for audio streaming to selected devices
Gateways (multicast only):
DT-24+No
6608Not until Seaview 3.3
VG200 (H.323)Yes with Cisco IOS. Software Release 12.2(11)T and later, enable the ccm-manager music-on-hold command
VG200 (MGCP)Yes, same as VG200 (H.323)
VG248Yes
Gateways (unicast only)AT-2/-4/-8, AS-2/-4/-8, and all other Cisco IOS. VoIP gateways, including Cisco 1750, 2600/3600, 5300, 58xx, and 72xx
Cisco IP phones (unicast, multicast)7910, 7940, and 7960
Cisco IP phones (unicast only)Cisco 7935, 12 SP+, and 30 VIP phones
Cisco IP SoftPhone (unicast only)
Maximum 250 simultaneous on-hold streaming sessions per server
Multiple server instances for application scalability
Multiple server instances for server load balancing and redundancy
G.711, G.729A, and wide-band audio codec support
Off-line audio translation utility
This URL should help you:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a0080094766.shtml

Changing Wave File Sampling rates

Hello,
I was wondering if there was a java class or package that can do this. I want to up sample a 8000 Hz wave file into a 44100 Hz one, and vice versa.
Any info would be appreciated!
Vicki

Use the javax.sound.sampled package.
Try this:
// Suppose you got a file, or an url
File file = new File("Blablablabla.wav");
AudioInputStream oldstream = AudioSystem.getAudioInputStream(file);
AudioInputStream newstream = AudioSystem.getAudioInputStream(
new AudioFormat(44100.0f,16,2,true,false),
oldstream
// Use the following code if you want to write to file:
AudioSystem.write(
newstream,
AudioFileFormat.Type.WAVE,
new File("blabla.wav")
);The variable oldstrea can also be obtained with an URL.
Check the part where the AudioFormat is constructed, you may want to change something, but if im right, this is cd quality.
The last method is for if you want to write the new stream in a file.
Note that if you change the file extension where it says new File("blabla.wav") and change the AudioFileFormat.Type you can also convert between wav and aiff and au etc.
R. Hollenstein

WAV files resynch every time iPod is connected despite no changes to files

Folks
I think the title pretty much explains my issue. Every time I plug my iPod Classic into iTunes, the software determines tracks to synch and re-synchs every WAV file I've imported and left checked. It has no problem with all 6,000+ mp3 files (they seem to synch fine), but repeatedly resynching the 200 or so WAV files I've got.
No fancy playlists or odd settings - iPod is pretty much stock and is set up to synch checked tracks only. Tried pretty much everything from uninstalling/reinstalling/repairing iTunes to hard reformatting and restoring the iPod. No change. It's almost as though the iPod is detecting changes to the WAV files on the laptop hard drive (like artwork has changed or something similar) and resynching every single time I reconnect.
Right now - and with only about 200 tracks - it's only taking about three or four minutes, but I'm converting most of my music to WAV format, so if it carries on like this the resynch time is going to increase exponentially . . . so I want to try and stop it now (and only synch updated tracks) before it gets out of hand.
Does this sound familiar to anyone else? Any ideas?
Thanks

Rusty Lantern wrote:
*Jun 29 09:58:07 Macintosh /System/Library/CoreServices/backupd[1218]: Event store UUIDs don't match for volume: Photos and Videos*
Hi, and welcome to the forums.
That means Time Machine isn't sure that what's on the drive is what it expects.
Jun 29 09:58:11 Macintosh /System/Library/CoreServices/backupd1218: Node requires deep traversal:/Volumes/Photos and Videos reason:kFSEDBEventFlagMustScanSubDirs|
Because of that, it must do a "deep traversal," comparing everything on the volume to the backups.
How are you disconnecting this drive? If you're not ejecting it first, while a backup is not running, that can corrupt the data on it.
Try a "full reset" of Time Machine, per #A4 in Time Machine - Troubleshooting (or use the link in *User Tips* at the top of this forum).
Also try repairing it, per #A5 in Troubleshooting.
See #A6 there for more detail on the messages.

Tempo problems with imported wav files

Hey everyone, sorry if there's a quick fix for this in the forums that I couldn't find, but I've been having some tempo problems with imported .wav files.
Long story short, my system couldn't handle playing all the tracks for a song while recording drums, so I bounced out an mp3 of the song and put it in a new Logic file so my drummer could just play along to that as I recorded him. Unfortunately, the original song is at 167 bpm, but I forgot to change the bpm in the new Logic file with the .mp3 file of the song to 167 bpm, so it was left at the default 120 bpm.
So, the drums were recorded at the correct 167 bpm, but Logic thinks that those new drum .wav files should be played at 120 bpm, so when I import my drum tracks back into the original file, they do not play correctly at all.
I could get record it all again, but I wanted to check if there was a way I could salvage what I already have, since my drummer lives about an hour away right now and can't just come over whenever he wants.
Thanks for the help! I really appreciate it.

Hi,
First, do not use MP3 in Logic, the sound quality is less than AIFF, WAV or CAF, and Logic has to decode it for playback, making it a heavier burden on the CPU than an uncoded audiofile, such as AIFF, WAV or CAF.
Secondly, audio files are independent of Logic's tempo. If you bounce down an audio file in any format (other than Apple Loop), it will play back at the same speed, +regardless of Logics' tempo setting+, either at recording or playback. Logic doesn't 'think' anything. The BPM is only important to MIDI tracks, or to the spacing between audio files. The audio files themselves *are not affected* by the tempo setting. If you import an audio file of tempo 167 into a 120 BPM project, the file will still play at 167, only Logic will indicate the wrong bar positions.
regards, Erik.

Read & Filter Multiple .wav files, Export Filtered SPL

Hi
I'm searching for tips on how to use LabVIEW to read multiple .wav files, then filter them, then export just the calculated SPL values to a "database friendly format".
Basically I want a solution that allows me to point to several files, then press a "go button" that triggers LabVIEW to produce filtered SPL values as exported data.
I'll skim over the cal approach since I'm confident most readers know how to handle it. Basically, though, for each 30 s .wav file of broadband noise, an associated cal tone file is used to determine the dBFS value (dB below Full Scale) to be associated with 94 dB SPL, or 114 dB SPL, as the case may be. I'll leave it at that.
Detailed Requirements, (excluding cal steps, tho).
1) Operates on up to a dozen 30 s time records (44,1 kSa/s, mono .wav files);
2) Passes the signal from each desired .wav file through 18 of the 1/3 octave filters (IEC 1260 compliant, as in the S/V toolset) not necessarily in real time!;
3) Calculates the SPL for each 1/3 octave filtered time record; This makes (12 files) x (18 filters) = 216 filtered SPL values if end-user points to a dozen broadband noise files before pressing the "Go" button;
4) Exports SPL values in a suitable database format; an existing database does the rest of the mathematics and report generation;
Client FileMaker database functions include:
- import and store 1/3 octave filtered Leq(30) SPL values;
- average filtered SPL across the declared qty of mic positions of the present test case;
NOTES:
1) In the field, we do NOT use simultaneous / MULTI-mics. Instead, it is absolutely an "open and shut case" that the optimum process is to move one (1) single mic quickly from spot to spot. While holding the airborne noise at a constant level we position the mic at each spot for 30 s to feed the .wav recorder, then move on. We emerge from a typical day of field activity with, say, 60 to 100 .wav files to post-process.
2) (1), above, explains why we don't use any N.I. DAQ h/w.
3) LabVIEW, it seems, exports best to a spreadsheet, so if needed, I'm prepared to handle manually any ugly conversion from LV's format to a proper database format ("1st Normal Form, 2nd Normal Form, etc.).
4) Given a choice to calculate the averages in my database, versus in LV (say, to cut down on exported data qty), I'd likely opt to export the 216 SPL's and then do averaging in the database, (where the ASTM E336 reports are done).
TIPS Gathered To Date:
1) S/V toolset has IEC 1260 compliant filters, BUT...
2) If you use the S/V toolset VI's, then you CANNOT ACCESS the filtered time record NOR CAN YOU WRITE the filtered data back to a .wav file! All you can do is measure the filters' outputs. Fortunately for us here, the available measures include SPL.
Aside: Although not a show-stopper in this project, this constraint is surprising, nonetheless. To access & record the filtered sound, you'd have to either (a) ask N.I. to unprotect VI's, or (b) develop your own IEC compliant filters from N.I.'s published tap co-eff's etc.;
3) LabVIEW does not support drag'n'drop operations on files. Instead, the recommended technique is to use multiple "Browse to/ Navigation/ File Open" panels to allow LV to open the user's desired qty of .wav files simultaneously;
Thanks for any tips!

The filter VI is password protected in the Sound & Vibration Toolkit (SVT) for two reasons. First, since the filter is compliant to a standard, by locking the code we guarantee that no one has the chance to accidentally change it so that they are no longer compliant. Second, the method of filtering the data is considered intellectual property that we are protecting.
That said, from reading your post it does not appear that you need to have access to the filtered data since you are making an SPL measurement which is provided in the Sound & Vibration functionality. Is that correct?
Also, I wanted to point you to another subVI which comes with SVT but is not in the pallette that might be of use. There is an example for SVT which is SVXMPL_Wav Power Spectrum which opens up a wav file and computes the power spectrum of it. Inside the VI, there is a subVI which will open a Wav file and convert it into a waveform which you can use with your SVT functions.
Hope that helps,
Jack

Why won't my .wav files show up in a 3rd party sound application after being exported as an omf file

Hi all, I've searched everywhere and can't find an answer, so hopefully someone out there will have mercy and help me!
I have Canon 7D footage (h.264 codec) and H4N Zoom audio (24bit 48khz .wav files) loaded into Premiere Pro CS 5. Editing on a Macbook Pro OS 10.6.8. I haven't added timecode.
My workflow, which I've done successfully several times in the past, is to sync the footage by creating individual sequences for each take (usually interview takes) and then bring in the audio and video and manually line/sync them up. I then link the two clips.
I then create a new sequence for my edit and then cut parts of the previously synced clips and then paste them into the sequence and edit away. After locking I export an omf (48khz, 24 bit, files as separate audio, aiff files, with 24frame handles). Then I import the OMF into Soundtrack Pro.
However this time, the majority of the sound clips do not show up in Soundtrack Pro timeline even though they are present in the omf folder as Aiff files. Normally you would think this was an issue with Soundtrack Pro, however one audio file did show up in Soundtrack Pro, but it was a sound file I had not linked to a video clip. Upon further investigation I found that that the audio clip that was not synced retained it's STE-014.wav name in the Premiere timeline. All the other audio I had synced to video had somehow taken on the video clip's MVI 0075 name.
I checked my old projects (the ones that I successfully exported to OMF) and even when I synced the audio and video, the audio retained it's STE.wav name. I don't know why the names changed; I did a test in my current project by cutting and pasting a synced clip at the end of the edit timeline and the audio retained it's STE.wav name. See pic.
After further tests, if the audio retains it's STE name it shows up in Soundtrack Pro (and Soundbooth as well), if it takes on the video MVI name it doesn't. I've tried exporting as an AAF and a Final Cut Pro XML file and it still doesn't work. I've tried unlinking and relinking the clips as well as making the audio offline and then relinking but still nada. I also cleaned the media cache to no avail.
If I render and replace the audio in the edit timeline, the name doesn't change, but the audio file does show up in Soundtrack Pro and Soundbooth, but unfortunately I need handles to edit the sound so this doesn't help me much.
Any ideas? and thanks!

Yes, it has always worked in the past. The past projects audio clips do not change names. I obviously did something different in this current project, but unfortunately I have no idea what.

ITunes can no longer read WAV files that QuickTime still can

Hello,
I don't known what happened, but iTunes doesn't want to read some of my WAV files anymore. The strangest thing is that QuickTime still does read them. But when I double-click on the track in iTunes, it moves immediately to the next one in the list.
Here are some of the details about the WAV files given by iTunes:
WAV
64kbps
16'000 kHZ
Mono
4-bits
DVI IMA
Thanks in advance for your insights

Yes, I its low-quality voice recording.
When I try to convert one of these WAV files to MP3, I get an error message that says that iTunes can either not read from or write to the disk.
Something must have changed in an update to iTunes. I was sent such a WAV file by email a while ago; at the time, I had managed to convert it to MP3 using iTunes. I dug up this WAV file from my email archive, and now iTunes cannot convert it to MP3 nor play it anymore.
Thanks for your insight.

Can no longer transfer .wav files to Zen Vision:M in Windows Media Player

Ok, first off let me state my goal so that if anyone knows a way for me to accomplish it without WMP that would be ideal:
I'm trying to copy lossless (wav) music files to my Zen Vision M including album art and all the information the player needs for artist, album, etc (rather than just plopped into the "all tracks" section.
Until today I was accomplishing this just fine in WMP. I would sync up my player and everything worked relati'vely well. Then today for some reason windows now wants to convert my wavs to WMA before copying to my zen. I don't know why and it's really irritating me. There must be a solution because it was working just a couple days ago on the same machine!
Thanks for your help

Well I figured it out. Turns out if you go to Sync(little down arrow)>DEVICE_NAME>Advanced Options>Quality if you leave it on automatic it will transfer wav files, but if you change it (I did without knowing) then the highest quality you're going to get is 92 WMA.

Problem in playing back mp3 which was recorded as wav file

Hi all,
I have recorded sound using my program. It uses classes like AudioSystem. And it's
able to record and play also. but in my site I have an mp3 player. So i thought of converting the recorded wav file into mp3. For this i used LAME utility in Linux. And it was converted. Also it is being played in normal players like Winamp. But when i try to play this converted mp3 file using my player in my site it's having problem. Why is it like that?
Should i do something while capturing the sound from microphone so that it can be converted into mp3 format??
Or
Can I directly capture voice to an mp3 file??
In my program,
targetDataLine.open(audioFormat) where audioFormat is like
float sampleRate = 8000.0F;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
Should I do something more or change something??
Please help me...
Thanx and Regards,
sand...

Thanks Walter.
Unfortunately, the same issue happens... In whichever window I select "Bounce" from -- Environment, Audio Mixer, or even FILE > BOUNCE -- a window titled "Bounce 'Output 1-2' " pops up. I'm guessing this is what you mean by the "Bounce Output Window." But, there I'm only able to name the file in the "Save As" field, choose where to bounce the file to on my harddrive, and then click on "Bounce" in the bottom-right of the window. I have no tabs or anything else to choose from. So I'm still a little stumped. Any additional thoughts?
I really appreciate the help.

CVP wave file change

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