I have schizophrenia since 1992. The following message is not directly relevant to my illness, but some answers from people here would be appreciated.
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.