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foobar2000 mobile / Re: Is it true that v1.4 can't even handle .ape?
Last post by Supermansaga -
Not sure, what you're talking about.
This APE file sample plays fine for me in 1.4.
TAK should also work, at least it did back in 2017 (provided via ffmpeg) according to the Android changelog.
Would have tested it as well but there don't seem to be any sample files on the net and I'm too tired to convert some on my own.

Sorry for not being clear. I was referring to the Android v1.4 version on my Pixel 7 Pro. No issue with iOS v1.4 on my iPhone 13 pro max. The problem occurred whenever i use ftp software to transfer the .ape and .tak files over from my windows 11. And they are 100% failure during transfer. So I thought to myself this must be due to the program not supporting these file formats to begin with. I guess I was mistaken in one way or another if now there are two people describing they can. But even though I have used another method of transfer called "nearby share" and the file lands in the path called "Internal Storage>Music>, it's still not visible on v1.4. So I must say something is still off here. 

Regarding .Tak, how do we add  ffmpeg to Android/iOS? Thx
Scientific Discussion / Re: What is the pre-amp input window range for dynamic microphone voltages
Last post by DVDdoug -
What is the pre-amp input window range for dynamic microphone voltages
It depends on the particular preamp.    Different preamps have different overload limits and different gains.

The most popular microphone of all time, the Shure SM57/58 puts-out 1.6mV at 1Pa (94dB SPL).    It's a dynamic mic, which is similar to a speaker.  It's a coil, magnet, and diaphragm...   An electrical generator.

Studio condenser mics have a built-in "head amp" which gets 48V "phantom power" from the interface or preamp.   Condenser mics tend to put-out 10 or 20 times more voltage than a dynamic mic.

Line Level (the output from a preamp) is about 1V so gains of 100-1000 are common.

With an interface we don't know the sensitivity of the ADC or the actual gain or output-voltage from the preamp. 

Most audio  interfaces are optimized for condenser mics and they may not have enough gain for a dynamic mic, depending on how loud the sound is.

I assume when I raise or lower the analog gain on my ADC I cannot amplify EVERY voltage (change) input from the microphone equally.
An amplifier is simply a linear voltage multiplier.     Digital amplification/attenuation is also done as multiplication.  Each sample (44100 samples per second, etc.) is amplified by the same factor (greater than 1 for amplification and less than one for attenuation).

Once the data is digitized/quantized there are "steps".    At 16-bits the data goes from −32,768 to +32,767 so the steps are very small.    With 8 bits you can only count to 255* (bigger steps) and if you make an 8-bit file the low-resolution results in audible quantization noise.   24-bits can hold values from −8,388,608 to 8,388,607.

0dBFS (zero decibels full-sale) is defined as the highest number you can "count to" with a given number of bits.   With floating point representation, a numerical value of 1.0 represents 0dB and for all practical purposes there are no upper or lower limits (as long as you remain in the digital domain).  The numbers in a 24-bit file are bigger than those in a 16-bit file but when you play the file, everything is automatically scaled to match the DAC (digital-to-analog converter).

Introduction To Digital Audio

Because of all that stuff above we can't talk about  :'( , I must pick a range to amplify it.  The min and max I effectively chose will determine how much noise or clipping I might experience.

There are two sources of noise.    There is acoustic noise in the room, and electrical noise from the preamp.   You can increase the acoustic signal-to-noise ratio by making the sound (signal) louder or getting closer to the microphone.  These will also improve the electrical signal-to-noise ratio.

The gain control in most preamps (including the preamps built-into interfaces) comes after the preamp so turning-down the gain control usually turns-down the signal and noise together so it doesn't affect quality.

Preamps have a voltage limit and they will clip if you try to go over.   Usually that voltage is high-enough that it "never happens".   The preamp built-into an interface has enough headroom so that the ADC (analog-to-digital converter) clips first.   It will always clip at exactly dBFS.   There is a direct correlation between dB SPL (and voltage levels from the mic) and dBFS but there is no standard calibration.  i.e.  If the SPL loudness goes-up by 3dB, the digital level will go up by 3dB (as long as you're not clipping.)

(If we do use dbs, and the mic has 60db of sensitivity and our Audio Interface has 120db dynamic range--wait, how does that make sense?  We shouldn't need a gain knob at all?)

Microphone sensitivity is given as a voltage at a specified SPL level.    94dB SPL (1 Pascal) is the "standard".

Digital recording levels aren't critical and pros often record at -12 to -18dB (at 24-bits).   The main thing is that you don't want to clip!!!   Low levels are often an indication of an analog problem and you might have a poor signal-to-noise ratio and the noise can become a problem when you amplify to "normal levels".   But turning-down the knob on your interface doesn't hurt anything as long as you have a good signal-to-noise ratio.

At very-low digital levels the quantization noise becomes a problem (again when amplified).     16-bits has a dynamic range of about 96dB and if you go below that, the "numbers" are zero and you have pure digital silence.    24-bits has a dynamic range of about 144dB.

* 8-bit WAV files are biased/offset and there are no negative values.    Of course, that bias is removed with it's played.

foobar2000 mobile / Re: 24bit/48Khz FLAC bug report from Playstore review
Last post by Supermansaga -
Someone reported that 24bit/48Khz FLACs don't play on his Android 13 device.
I tried this file and it plays on my Android 13 smartphone. Does anyone else have problems, probably with different files?

If we are supposed to hear noise from your file, then it works for my Pixel 7 Pro which has Android 13 right out of the gate. That someone had best shared their file to test.
3rd Party Plugins - (fb2k) / Re: Game Emu Player (GEP) not working in foobar2000 2.0
Last post by Chinō -
That certainly is a strange problem. I can't offer a solution to it per se, but I recommend installing foo_input_vgm, as it is more up-to-date and feature-rich. Hopefully it would fix your problem.

ACTUALLY, IT WORKED! I just have to select the correct sound chip to mute the channels freely! Thank you so much! Have a great day!
FLAC / Re: FLAC v1.4.x Performance Tests
Last post by Porcus -
8192 beats 16384 size-weighted in my tests on upsampling those 38 CDs, but it varies quite a lot. Medians could very well tell a different story. I messed up something and it is still running, but I can report on those two block sizes at least.

Using -A "subdivide_tukey(3);blackman" on everything, then
** In overall size:
At 96 kHz, -b8192 beats -b16384, and -eb8192 beats -eb16384.
At 192 kHz, same happens.
At 384 kHz, -b8192 beats -b16384 by around 0.12 percent, but -eb16384 beats -eb8192 by around 0.18 percent.

192 kHz, let's look into that further: No -e here.
* Classical music benefits from larger blocksize -b16384, 12 albums to 2; all except harpsichord and (near-zero) Cage's percussion works. Total impact 0.32 percent (not percentage points!), varying from -0.15 (harpsichord) to 0.63 percent (Bruckner, vocals)
Median impact = 0.37 = median absolute value impact.
But then the rest:
* The heavier music: -b8192 wins by 7 albums against 3, switching sign on impact to signify that:
Total impact -0.14 percent, varying from -0.71 (Laibach, biggest benefit for -b8192) to 0.24 percent (Gojira, that benefits from 16384).
Median impact = -0.24. Remove the sign for median absolute value impact.
* The others. -b8192 wins by 9 albums against 5
Total impact -0.28 percent, max benefit from -b8192 is -1.31 percent (Wovenhand, in this release that is singer/songwriter) and then -0.99 (Sopor Aeternus, that is something completely different: darkwave) - and on the other end, benefiting most from larger blocksizes are the jazz albums: 0.41 percent for both Davis and Johansson. Those were near-mono before dithering I think.
Median impact = -0.32 percent. Median absolute impact: 0.38.

For those who did not follow the previous discussions, I am talking about optimizations for >=4x upsampled data, so the parameters listed above are not suitable for encoding "real" hi-res files.

... but who knows how many hi-res files are "real".