2013/2/9 Erik de Castro Lopo <mle...@mega-nerd.com>: > Johnny Rosenberg wrote: > >> For example, when loading an audiofile with libsndfile, all the >> samples are converted to floating point numbers, > > With libsndfile, converting to float is optional. You can just as > easily read int or short. > >> which is what I want >> anyway, but the conversion is done by dividing the integers from the >> file with pow(2,bps-1), but when converting back, they multiply with >> pow(2,bps-1)-1, so if you just read and then write, you end up with a >> slightly lower volume (not that you can hear any difference if you do >> it only a few times, but still, it isn't right, AND it is very easy to >> do it right, so why don't they?)… > > Firstly, you do realise that I am the main author and maintainer of > libsndfile, don't you?
Yes I do, now that you mentioned it. I'm not sure why that matters in a FLAC-dev mailing list, though. I didn't know there was a maintainer anyway these days, since I didn't get a reply when emailing about this a couple of years ago (or maybe it was last year, I don't remember – I sent it from one of my other email addresses, since I use this one for mailing lists only). > > Secondly, the scaling can be switched off don't you? See: > > http://www.mega-nerd.com/libsndfile/command.html#SFC_SET_NORM_DOUBLE Thanks, I don't think I saw that page before, for some strange reason. The sentence ”setting normalisation to SF_FALSE means that no scaling will take place”, does this mean, for instance for a 24-bit file, that the values will be doubles in the interval -8388608 to +8388607? That information should probably be added to question 10 in your FAQ, I suppose. For my purpose, not ”normalising” at all seems to be a good enough idea, I think. I will calculate the levels from dB anyway, so I just need to know how many bits per sample there is, which libsndfile easily lets me know. But I will try to do my thing with libflac first. If I fail making my code clean, simple and readable (and working…), I might give libsndfile another try. > > Finally, there are about 700 different way so convert between int > and float. I chose the one that I thought had provided the best > trade off. I stand by that decision. Well, the good thing is that this SFC_SET_NORM_DOUBLE thing allows me to do it the 701th way… Maybe I should convert to % or ‰… Anyway, if I'm allowed to ask a libsndfile question here, is there a special reason why you read sound files into one-dimensional arrays instead of two-dimensional ones (such as AudioData[Channel][Sample])? If so, what is that reason? Thanks. Kind regards Johnny Rosenberg ジョニー・ローゼンバーグ > > Erik > -- > ---------------------------------------------------------------------- > Erik de Castro Lopo > http://www.mega-nerd.com/ > _______________________________________________ > flac-dev mailing list > flac-dev@xiph.org > http://lists.xiph.org/mailman/listinfo/flac-dev _______________________________________________ flac-dev mailing list flac-dev@xiph.org http://lists.xiph.org/mailman/listinfo/flac-dev
