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Hello everyone, everything good?

Can anyone with experience help with the following question?

Using Free Audio Converter 1.0.30 what is the best FAAC configuration 1.29.7 using bit rate per channel without Joint-Stereo?

128, 130, 150 ....

Because in the version I used 1.0.28 with FAAC 1.28 I used 128 per channel and the songs got the final quality this way:

247kbps, 265, 270 ....

And with FAAC 1.29.7 when choosing which rate per channel the songs have the same qualities for example:

130 per channel result = 260kbps.

What is the best bitrate per channel setting in FAAC 1.29.7 without Joint-Stereo?

I apologize for extending the text too much.

Thank you!
WavPack / Re: bits/sample, kbits/second?
Last post by [JAZ] -
Ouch... I needed to check on which subforum this post was in, to even understand what the question was about.

From the manual
-bn = enable hybrid compression, n = 2.0 to 23.9 bits/sample, or
                                 n = 24-9600 kbits/second (kbps)
The default operation of WavPack is pure lossless, which means that the .wv file contains all the information that was in the original .wav file. The hybrid mode allows the user to specify a target bitrate for the output file, either in kilobits per second (kbps) or bits per sample. If the track can be losslessly compressed without exceeding the specified bitrate, then it will be and WavPack will report the compression as lossless. If lossless compression would exceed the specified bitrate, then WavPack will begin carefully discarding the least significant portion of the audio information to stay within the limit. Every effort is made to keep this inaudible, including the use of joint stereo, dynamic bit allocation and noise shaping. WavPack will report this as "lossy" compression. Although the option accepts bitrates as low as 24 kbps, the actual value that WavPack can achieve is usually much higher than that. For example, with CD-audio sampled at 44.1k the lower limit is about 196 kbps.
The hybrid mode can be used quite successfully with floating-point audio, however it should not be used for scientific type floating-point data because the hybrid algorithm might not be application appropriate (and floating-point "exception" values like infinities or NaNs will not be properly encoded). Use only the pure lossless mode with non-audio floating-point data.

So, first, we are talking about the lossy part of hybrid encoding of WavPack.

I don't know the innerworkings, so I will not give the best answer, but basically, you tell the encoder to constraint the quality either by the desired signal to noise ratio, or by the desired average bitrate of the lossy file.

The bitdepth (bits/sample) determines the allowed amount of noise distortion on the signal. At 24bits (for a 24bit file) it doesn't allow any distortion. At 16bits, it would allow a distortion that would be around the noise floor of an Audio CD.  8bits would be a noisy background, but where music could still be heard without much issues (except if it is a low volume passage).

The bitrate (kbps/second) determines the desired size of the file ( given a bitrate and a duration in seconds, you get the file size by multiplying those values). In this case, the encoder is responsible to determine the allowed noise floor, i.e. the allowed bitdepth.

I don't have enough knowledge of the codec to say if the bitrate mode can change the bitdepth dynamically, or it is more or less fixed, but I remember that when LossyWav was developed, that Wavpack was not so advanced.
Audio Hardware / Re: EQing headphones
Last post by Arnold B. Krueger -
As those of you who have followed this column for any length of time can attest, headphone mixing is one of the big no-no's around these parts.  In our humble opinion, headphone mixes do not translate well in the real world, period, end of story.  Other than checking for balance issues and the occasional hunting down of little details, they are tools best left for the tracking process.

Yeah sure....I guess that is why most of the new music out there sounds like sh*t. Use both!

Hostility towards headphone mixing is hardly a new thing. I remember receiving this advice decades ago. I have no use for it.

How headphones work for a person probably has a lot to do with personal experience. I've been listening criticallyfor extended periods of time,  to music longer than I have been licensed to drive which is over 50 years. I suspect that an effective translation from headphones to speakers and the real world  got wired into my brain early on. 

Consequently I am not infrequently told that my headphone mixes sound good on speakers. 

When I listen to a well-placed coincident pair of cardiod mics, the sonic scene I hear translates into the position of the performers that I see. Their may be some auto suggestion in that, but I use that info to place microphones and it seems to very frequently work out.

I have no idea why some people demand cross-feed circuits for headphone listening.

Actually, I can kinda-sorta translate from how headphones sound to me to how others may hear them and then I can kinda-sorta hear why people may want crossfeed.
WavPack / bits/sample, kbits/second?
Last post by Valsu -
I'm interested in the advantages/disadvantages of using -b with a value smaller than 24 (bits/sample) vs. one bigger or equal to 24 (kbits/second). What's the technical difference and if there is any, how do these two "modes" differ with regards to quality?
Audio Hardware / Re: EQing headphones
Last post by DVDdoug -
Yeah sure....I guess that is why most of the new music out there sounds like sh*t. Use both!
Of course, use both...  Pro mastering & mixing engineers check their mix on a variety of systems (after doing most of their work with monitors).   If you're an amateur without experience, good monitors, or an equalized room, you should check your mix on everything you can get your hands on!

BTW - That advice comes from Recording Magazine's Reader's Tapes Column where amateurs & musicians send in there mixes to be critiqued.     After hundreds or thousands of submissions, people monitoring with headphones generally get poor results (or poorer results).    And of course, pros use monitors.

The exception would be if you are making a binaural recording or other special recording intended for headphone listening.
You are welcome. I sometimes wonder whether all these flaws could have been avoided had the mp3 file format been properly designed from the outset, or whether it is inevitable from the ubiquity of mp3 that all sorts of nonsense are to be found in the wild.
Nothing has changed regarding --no-delay option.
--no-delay is safe as long as the beginning (50ms or so) of input signal is digitally silent.
Moreover, even when the beginning of input signal is not silent, you won't hear any "glitches" as long as you listen to one file only.
The effect of --no-delay should become apparent only when you encode two or more gapless inputs, and you listen to the gapless transition between songs.

Alright thanks for all !
3rd Party Plugins - (fb2k) / Re: foo_discogs
Last post by eamatag71 -
Now it's a mystery....
3rd Party Plugins - (fb2k) / v1.0.2
Last post by TheQwertiest -
Version: 1.0.2
    • Added foo_title to Preferences -> Components menu. Thus it's possible to view foo_title's version from the same menu as other components.
    • Fixed bug: 'enable when minimized' was not working at all.
    • Fixed inconsistent fade-in/fade-out animations (hopefully for the last time).
General - (fb2k) / Folders
Last post by cmg -

It may be that this question has already been asked but I have not found the answer.
I use Foobar mainly to convert music files and the folder where I have the files to be converted is D:\Musics, but, if the last file Foobar used is no longer in that folder, when I open the program, invariably the opening folder is something like c:\users\docs\.... and never D:\Musics.
How to set foobar to open in this folder have or do not have any file?
Thanks in advance