Audition 5.5 - Help with .wav files for phone system - making a hold message PC & Mcc

Hello everyone,
I am really frustrated trying to create a .wav file for our phone system and I was hoping someone might have some suggestions as I am new to using Adobe Audition. Our phone system will only accept .wav files for our hold messages but the tricky part is that is can only be a maxiimum streaming bit rate of 64 kbps. Everything we've created under 256 kbps becomes in-audible and I'm really not sure what to do as the current messages sound fantastic. Our current hold message is 35 seconds long and we have another one we are about to do which will be 2 minutes long (uuuugh).
- I've tried first converting the .wav file to an .mp3, lowering the bit rate to 64 kbps, then converting back to .wav but the bit rate goes up to almost 1500 kbps after re-converting.
- I've tried making the track mono, reducing the sample rate to 22k but the size ends up being somewhere between 512 and 768. If I lower the sample rate much further and convert to 8bit, the audio sounds terrible.
- I've also tried using Adobe Media Encoder, Audacity, Hand Brake, OS X Automator and a few other programs all with the same results.
- I've tried re-recording the audio on a handheld .mp3 recorder hoping the initial quality would be less (originally recorded on a PC through Audition with a external mic).
Unfortuantely, the person who used to do these conversions is no longer with the company, never documented the process and all I know is that they used to do this on a Mac. Also, the software that uplaods the files is a Linux based which is out of my area of expertise so I'm not able to tinker with it to make any headway.
** Any sort of suggestion would be greatly aprpeciated as I really don't know much about audio. I've been working on this for 2 weeks now and this is really driving me crazy. **

9ov7 wrote:
Hello everyone,
I am really frustrated trying to create a .wav file for our phone system and I was hoping someone might have some suggestions as I am new to using Adobe Audition. Our phone system will only accept .wav files for our hold messages but the tricky part is that is can only be a maxiimum streaming bit rate of 64 kbps. Everything we've created under 256 kbps becomes in-audible and I'm really not sure what to do as the current messages sound fantastic. Our current hold message is 35 seconds long and we have another one we are about to do which will be 2 minutes long (uuuugh).
- I've tried first converting the .wav file to an .mp3, lowering the bit rate to 64 kbps, then converting back to .wav but the bit rate goes up to almost 1500 kbps after re-converting.
Eugh! Converting to MP3 is absolutely not the thing to do!
- I've tried making the track mono, reducing the sample rate to 22k but the size ends up being somewhere between 512 and 768. If I lower the sample rate much further and convert to 8bit, the audio sounds terrible.
- I've tried re-recording the audio on a handheld .mp3 recorder hoping the initial quality would be less (originally recorded on a PC through Audition with a external mic).
Okay, in reverse order: I know it sounds daft, but actually you want to start with the best recording you can possibly have. You are going to have to reduce its bit rate, but if you reduce a crap signal's bit rate, it inevitably ends up worse than reducing a good signal's one; when you think about that for a moment, it's going to make sense (I hope...).
As for what you need to do - well it's going to have to be a mono file anyway (you haven't got a stereo telephone system), and that will immediately halve the streaming rate, but the other thing you need to bear in mind is that telephones have a limitited bandwidth, and that your sample rate, even at 22k, is going to give you more bandwidth than the system has. If you can reduce it to 8k, it will work fine. This will still give you an upper frequency limit of 4kHz, and that's more than adequate for telephone speech, believe it or not...
The other thing, and this is where I think you may be running into trouble, is that 8 bit streaming is pretty much the norm here. All it does is to increase the noise floor, so that the signal to noise ratio is about 48dB instead of the 96dB it would be with a 16-bit signal. But, to get this to work to your advantage, you have to optimise the audio you are using in the first place. Chances are that you need to compress your recording so that it has a relatively limited dynamic range before reducing it to 8-bit, and then you'll be fine. This is better anyway for telephone systems - there's far less chance of bits of the message being missed because they're too quiet. Also, you don't want massive great peaks in the recording - these just reduce the maximum overall level for the majority of the message sound. So, you need to apply a limiter to get rid of them, leaving the majority of the speech sounding just as good, only louder. All of this you can easily do in Audition - don't need any other software at all.
So, my guess is that the previous maker of your messages knew exactly what he was doing. Yes, he could have documented the basic steps, but he couldn't have been absolutely specific about it because no two recordings are the same. I'll attempt to explain the basic steps in English, but how well you will be able to follow this, I'm not sure...
The basic process I'd go through with the recordings is firstly to normalize them to 0dB, then look at how far down most of the 'green' parts are - this is the bulk of the signal, and where all of the meaning is. Then, you apply a limiter to this so that those green parts don't have peaks sticking out of them. Then, if you need to, normalize again so that the green part moves to the upper part of the waveform space (towards where 0dB is). Then, if you need to, apply compression to the signal so that the level of it sounds about the same all through the message.
Now, when you reduce your signal to mono, 8k, and 8-bit, pretty much everything about the original signal will still be there. That should stream 8-bit words at 8k, making 64k streaming for a single mono channel. This will work fine, but only if you optimise the signal first - otherwise no chance!

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