Hi all, I start this thread in hope to gain/share some knowledge about music samples and working with them.
I'm not interested in music mixing or making songs, more rebuilding and remastering old songs and music samples.
First off, I use Linux, and at the moment only use open source software. I have come across many applications that seem to offer a wide range of tools but most are geared tword mixing or have very few filters for dealing with the basis of the sample itself such as actual bit-rate and frequency.
I will explain my problems, My father backed up his music collection a couple of years back from his damaged CD's using the FLAC container.
In total there is about 3gb of music which sounds awfull, but, I find that if I decompress into WAV the sound is improved tenfold, but so is the size.
3gbites of FLAC has now become over 70gbites which is just too large to be practical with songs that were ~12mb in size now being ~120mb?
The sound quality difference has me baffled, listening to one particular song in WAV format I can clearly hear instruments and notes that were not there when played in compressed format, I understand how frequency's work, i dont understand how uncompressing the song has results like this, as if some act of magic has happened and notes and instruments just appear!...
If this is the case with compressed formats I also do not understand why media players dont uncompress or buffer songs as they play?
My main question here is about actual bit-rate/frequency, How does one tell the actual rate of a sample and not just what the container reads it to be?
If I understood this I could go ahead and compress these files back to the origional recorded bit-rate.
Suggestions/thoughts please.
I'm not interested in music mixing or making songs, more rebuilding and remastering old songs and music samples.
First off, I use Linux, and at the moment only use open source software. I have come across many applications that seem to offer a wide range of tools but most are geared tword mixing or have very few filters for dealing with the basis of the sample itself such as actual bit-rate and frequency.
I will explain my problems, My father backed up his music collection a couple of years back from his damaged CD's using the FLAC container.
In total there is about 3gb of music which sounds awfull, but, I find that if I decompress into WAV the sound is improved tenfold, but so is the size.
3gbites of FLAC has now become over 70gbites which is just too large to be practical with songs that were ~12mb in size now being ~120mb?
The sound quality difference has me baffled, listening to one particular song in WAV format I can clearly hear instruments and notes that were not there when played in compressed format, I understand how frequency's work, i dont understand how uncompressing the song has results like this, as if some act of magic has happened and notes and instruments just appear!...
If this is the case with compressed formats I also do not understand why media players dont uncompress or buffer songs as they play?
My main question here is about actual bit-rate/frequency, How does one tell the actual rate of a sample and not just what the container reads it to be?
If I understood this I could go ahead and compress these files back to the origional recorded bit-rate.
Suggestions/thoughts please.
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