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This message follows on from a previous message to this forum. I need
confirmation that I have used the right parameters saving to MP3, and
that the audio quality of the MP3 file is as good as that of the WAV
file.

This message pertains to; http://c-compiler.com/myfiles/a-mp3.zip

The original WAV is at; x.wav

I have converted this file to MP3. Please listen to; x.mp3

According to Windows Media Player, the bit rate of the source file is
192 Kbps, see "windows-media-player.jpg"

According to VLC, the source file has sample rate 48,000 Hz and bits
per sample 16. This particular codec (IMA WAV ADPCM Audio) actually
has 4 bits per sample, but this is decompressed to 16 bits per sample.
See "vlc.jpg".

According to MediaInfo, the source file has sample rate 48,000 Hz and
bit rate 192 kb/s, with a bit depth of 4 bits (which is decompressed
to 16 bits, as noted above), see "mediainfo.jpg".

According to Total Recorder, the source file has sample rate 48,000 Hz
and bit depth 4 bits, see "totalrecorder.jpg".

I use the LAME encoder with Total Recorder to convert the WAV to MP3,
see "totalrecorderA.jpg".

The media format in Total Recorder specifies sample rate 48,000 Hz and
bit rate 192. This is in keeping with the parameters for the source
WAV file, see "totalrecorderB.jpg".

Finally, opening the new MP3 file (converted from WAV) gives the
screen shown in "totalrecorderC.jpg". Bit rate for the MP3 is 192
kbit/s and sample rate is 48,000 Hz.


There are essentially two questions I need to ask.

(1) I have used the parameters for the source WAV file when creating
the MP3 file. Is this a sensible approach? Audio quality is top
priority.

(2) Please tell me if the audio in the MP3 file is as clear as with
the WAV. I think it is, but I would like to be re-assured.

The words on the recording are, "people like that should be .... I
know, they should be homeless".

Thank you for responses.