I am a noob to signal processing, but I need to create some audio files for IVR for Asterisk VOIP.
So my lady friend with a nice voice records her message on a Windows machine and sends me a 44KHz, stereo 32bit wav file.
I currently squash it into a 8KHz, mono, 16bit signed wav file for Linux, and then convert the wav to alaw or ulaw PCM format.
One source of info for this process is this link.
I can do some magic with Audacity:
- Stereo to mono
- Equalize, using a "voip" or "telephone" filter: discard anything over 3KHz and under 300Hz.
- compress volume/levels to -12 to -15 dB
- Resample 44 -> 8KHz
- Export as 16bit signed PCM wav
- Move it to Linux and sox it to alaw or ulaw
The result is acceptable, but I feel it could be improved. In particular, after resampling, the voice sounds muffled.
Is this order correct?
Do you have suggestions to get better results?