Correct MP3 parameters for minimize lossy-ness and file size

Hi folks,
Correct MP3 parameters for minimize lossy-ness so as to miminize file size.
I am given a voice recording file in 32k sample rate and bit depth = 16. On spectral frequency display I see that there is not much useful data beyond 3200Hz, which I can ignore.
If I change the sample rate to 11025, frequency range is about 5000Hz which adequately holds the valid range. Now my question is do I lose anything by downsampling the data? Does the downsampling alters some data in the 0-4000Hz range?
My second question is the default option while saving in .mp3 for a file with 11025 sample rate would be 32 Kbps, 11025 Hz. do I gain any quality by encoding to a higher say in 40Kbps or more? What is a best MP3 encoding option for voice only recordings.
Is MP3 more lossy than ACM/ADPCM?
Another off topic question-- Can I find a particular sound? Suppose I see a particular noise and selects that and take corrective measures now I would like to find out any other occurances of that type of noise in the whole file. I understand that exact match is not possible but can I search to find out somewhat similar sounds?
best wishes to you all,
Deeptish

If I change the sample rate to 11025, frequency range is about 5000Hz which adequately holds the valid range. Now my question is do I lose anything by downsampling the data? Does the downsampling alters some data in the 0-4000Hz range?
In theory, no. If you use a lousy resampling method you will achieve some intermodulation distortion, however Auditions sample rate conversion process is pretty on the dot.
Is MP3 more lossy than ACM/ADPCM?
Yes/No. MP3 and ADPCM (adaptive differential pulse-code modulation) are two vastly different technologies (incidently, the ACM simply stands for Audio Compression Manager, and simply means it's an ADPCM codec for Windows). ADPCM is based off PCM and works in the time-domain. It uses dynamic compression to reduce the dynamic range so the audio is able to be mapped to an 8-bit resolution without excessive quantization noise. Upon playback/decoding, the signal is expanded. The process is far from lossless, however, A-law and μ-law have been telecom stanards of ADPCM for carrying voice traffic for years.
My second question is the default option while saving in .mp3 for a file with 11025 sample rate would be 32 Kbps, 11025 Hz. do I gain any quality by encoding to a higher say in 40Kbps or more? What is a best MP3 encoding option for voice only recordings.
MP3 is an entirely different game. Unlike ADPCM which works on the signal in the time-domain...MP3 processes an stores it's audio in the frequency domain and using physcoacoustic models, removes information from the frequency domain to reduce the number of bits needed to store the audio. The frequency response of an MP3 is determined more by the bitrate than the samplerate. A MP3 that's about 55kbps VBR sampled at 22khz only has a frequency response of 7.5khz, however, that doesn't mean the audio less than 7.5 khz hasn't been modified. Physcoacoustics work over the entire signal, the higher frequencies are the first to go since physcoacoustics figure we'll miss them less...but it does remove audio from the signal in many other bands as well. Sample rate does play a role...some bitrates are not available for specific sampling rates; since a 64kbps file might have 8khz of bandwidth, it's deemeed "too much" for 11khz and is not a valid rate. A 128kbps mp3 using 44.1 and a 128kbps mp3 using 32khz will generally have about the same range of spectral response....save for newer encoders that can squeeze a couple of muddy khz in there. A 32kbps mp3, in practice, is acceptable for voice, there should be about 4khz of available bandwith, higher bitrates will naturally give you better quality.
Basically, either MP3 or ADPCM are lossy formats, they're just radically different implmentations of audio compression with thier own sets of advantages and disadvantages.

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Chemistry guy wrote:
> I'm new to Flash, and I'm trying to decide how to set
the bit rate (kbps) for
> my audio files.
>
> I'm creating PowperPoint-like presentations with voice
audio. You can see what
> I'm doing at
>
>
http://preparatorychemistry.com/Bishop_Audio_Book.htm
>
> I first decided on 16 kbps, but I'm not happy with how
it sounds, especially
> through headphones connected to the computer. My
presentations, of course,
> sound better with higher bit rates, but file sizes
roughly double for each
> doubling of the bit rate. At 16 kbps, my files range
from 1-3 MB, so for 32
> kbps (which I'm leaning toward now), they are 2-6 MB. My
questions are
>
> What's the concensus on the best bit rate for Internet
sound files, especially
> voice?
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> How important is it to minimize files size? The files
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> so that dial-up folks will probably not listen to them,
and even the larger
> files seem to download fast enough with my cable
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<urami>
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