The devil in the detail is hardly ever mentioned, and most users will anyway never encounter that mismatch.
ders don't "eat" audio information and silently leave on the plate what they don't like. Every stream has a format and every encoder a set of formats it accepts. For your case, a developer would have to intentionally convert the data before feeding it and intentionally not inform the user about it. That is very unlikely ...
With foobar2000 1.1.1 and flac encoder 1.2.1b:Source file = wav 32-bit (floating-point)Output file format = flacOutput bit depth = either "32-bit" or "Auto"Due to the specifications of the flac format two identical 24-bit flac files are created, without an error message or a lead on the bit depth reduction. foobar's console reports: "Track converted successfully."Conclusions:Porcus posed a legitimate question.It's the duty of the user to check which specifications are supported by the encoder he uses.
Since floating point numbers are by definition approximate, I don't think theres an expectation that processing will be lossless.
All tracks decoded fine, no differences found.
Does that result not mean that a lossless conversion of a floating-point file can be expected or is there an error in my logic?
Then, arrgh, decoder bugs: http://www.hydrogenaudio.org/forums/index....showtopic=84101
I still think that Foobar or the Flac encoder is far from behaving as it should
Float preservation is hard to accomplish. It's not impossible, though. Developers must just not rely on a specific hardware's float capabilities for the correctness of their program.
I think this too, but why would it be wrong to mention in the FLAC Wiki that this file format can have 24-bit max, but accepts 32-bit sources as input files and that the encoder doesn't provide a warning in that cases?
I admit that I don't understand that part of your post starting with "this is where the information loss occurs (not in the encoder) - and the developer is not telling you about it!"
If the information loss doesn't occur in the encoder: whose developer do you mean then with "and the developer is not telling you about it"?
So I would like to hear Bryant's (David's) comment on this. I am using several computers (= different hardware) and converting mainly 32-bit floating-point wav files to WavPack and on all these PCs there never has been found a difference between an original source file and a WavPack file that I converted back.
Floating point is only approximate
FLAC doesn't convert 32-bit input to 24-bit. Foobar2000 does this.
Edit: lvqcl has already provided the answer.
f32.wav: ERROR: unsupported format type 3
i32.wav: WARNING: legacy WAVE file has format type 1 but bits-per-sample=32i32.wav: ERROR: unsupported bits-per-sample 32
i32w.wav: ERROR: unsupported bits-per-sample 32
By the way, WavPack accepts 32-bit integer files, but foobar2000 converts 32-bit integers to 32-bit floats.
The devil in the detail is hardly ever mentioned