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Partner Messaging and wav files? 3
- Thread starter allisland
- Start date
deepblue63
IS-IT--Management
I would be interested in this also...
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3
- #3
Hello! My first post, and a big thank you for all the previous info I've received here. It was instrumental in getting R7 up and running. The modem info was perfect timing!
I've made several recordings using programs like CoolEdit2000 and Adobe Audition (the new CoolEdit2K). The Avaya docs I've read indicate the sample rate ends up as 8KHz at 8 bits. Most know this is the typical bandwidth of analog telephone lines. Here's what I did - the tools you use depend on how "large" sounding the resulting message is.
I did all the prep work using CD quality rates (44.1kHz at 16 bits). I mixed it down to 11kHz @8bits. I actually used background music beneath the voice.
The bottom line:
Save the format as Windows PCM.
Voice+Music: Save as 11,000Hz at 8 bits.
Plain Voice: Save as 11,000Hz at 8 bits, then "compress".
Here is where the tools make the difference. CoolEdit2K and Adobe Audition have lots of features, one of which is Dynamic Processing. This allows you to compress the resulting speech (before downsampling). This process equalizes the speech so it attempts to be at a constant amplitude (equal volume). This is not required, but the sound is optimal sounding over the phone. Obviously, if you have the tools to play around you can make it sound better using a mic rather than the telephone.
If you try to send the system a file with a high sample rate it becomes too decimated and it sounds fuzzy with a low amplitude.
Sometimes you may get an error (while uploading the file) that it needs correction. Say YES, and the system will fix it. Using a sample rate of 11kHz is just high enough so after the system processes the WAV file, you get the best quality (IMHO). I've tried it at 8kHz and have noticed a slight reduction in quality.
Hope this info helps!
I've made several recordings using programs like CoolEdit2000 and Adobe Audition (the new CoolEdit2K). The Avaya docs I've read indicate the sample rate ends up as 8KHz at 8 bits. Most know this is the typical bandwidth of analog telephone lines. Here's what I did - the tools you use depend on how "large" sounding the resulting message is.
I did all the prep work using CD quality rates (44.1kHz at 16 bits). I mixed it down to 11kHz @8bits. I actually used background music beneath the voice.
The bottom line:
Save the format as Windows PCM.
Voice+Music: Save as 11,000Hz at 8 bits.
Plain Voice: Save as 11,000Hz at 8 bits, then "compress".
Here is where the tools make the difference. CoolEdit2K and Adobe Audition have lots of features, one of which is Dynamic Processing. This allows you to compress the resulting speech (before downsampling). This process equalizes the speech so it attempts to be at a constant amplitude (equal volume). This is not required, but the sound is optimal sounding over the phone. Obviously, if you have the tools to play around you can make it sound better using a mic rather than the telephone.
If you try to send the system a file with a high sample rate it becomes too decimated and it sounds fuzzy with a low amplitude.
Sometimes you may get an error (while uploading the file) that it needs correction. Say YES, and the system will fix it. Using a sample rate of 11kHz is just high enough so after the system processes the WAV file, you get the best quality (IMHO). I've tried it at 8kHz and have noticed a slight reduction in quality.
Hope this info helps!
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