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French | |
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Author | Message |
Waynesail
Newbie ![]() Joined: 10 October 2007 Online Status: Offline Posts: 1 |
![]() ![]() ![]() Posted: 10 October 2007 at 7:31pm |
These FSI files are an excellent resource. However, I noticed that they are a little hard to hear clearly, both on my home computer with no background noise, and even more difficult on my mp3 player. The sound has a muffled quality. I see that they are digitized at 32kbps. It has been my experience that 64kpbs is the minimum necessary (for me, at least) to clearly hear an mp3 file.
Is anyone else experiencing any difficulty?
Does anyone know if the files are a home digitization of tapes? Or were they originally created at FSI at only 32 kbps?
I'm wondering whether anyone has the files digitized at a higher bit rate that you would be willing to post?
Thanks,
Waynesail
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DemiPuppet
Administrator ![]() Joined: 27 May 2006 Location: United States Online Status: Offline Posts: 163 |
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The original source of the audio comes from cassette tapes. For the audio I supplied I did the following:
1. Digitized the signal by playing the tape with a good Technics cassette player into M-audio FireWire AD converter at 44.1 KHz sample rate. Every 15 tapes or so I would clean and de-magnitize the tape heads. 2. Save the recorded stereo waveform as a 16 bit PCM WAV file. 3. Convert the stereo WAV file into a mono WAV file (sum both channels and divide the amplitude in half). The assumption is that the 1960's era tapes were originally mono. Converting the stereo recording to mono reduces some of the difference noise between the channels. 4. Low pass filter with a sharp 8Khz corner frequency using a 256 sample window. The highest frequency sibilant in the human voice is about 8Khz. 5. Determine the peak amplitude a and then raise the amplitude of the whole file so that the peak value is 3dB below the absolute possible peak. This is done while digitally expanding the signal. Signals below about -30 db (hiss) will get considerably less amplitude increase than signals above the value. I did a mild 3dB reduction of any signal less than -50dB. I didn't do any more than -3dB because I found it too annoying; it would sound like someone is turning the volume up and down. I notice this on the French Basic audio files (I didn't digitize them). 6. Convert from 44.1KHz down to 22.05KHz (lossless since the signal was low passed filtered to below 11Khz). 7. Submit the lossless uncompressed WAV files to this site where they were then converted to 32 kbps MP3s. I don't know what encoder settings were used. |
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