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3
3rd Party Plugins - (fb2k) / Re: Playlist Organizer (aka foo_plorg) replacement on Jscript Panel 3
Last post by etip -
Playlist Organizer (Jscript Panel 3)
v2.0.4
03-10-2023

Version change :
- Added functionality to automatically create a (regular) playlist when songs are dropped into an empty space of the panel.
The playlist will be named `New playlist` or have numbers appended if there are existing playlists with the same name.

X

Ps : I will think of a mechanism to name the new playlist, as described below, but later.
4
Scientific Discussion / Re: What is the pre-amp input window range for dynamic microphone voltages
Last post by ktf -
There are some new 32-bit floating-point interfaces.   I assume that will be getting more common.    From what I recall, this gives you a digital range of around -1000dB to +1000 dB (infinite for all practical audio purposes) and Zoom claims "no need to set gain".   But there is still analog noise and analog clipping.   You might get 20dB more usable dynamic range with a floating point interface, but I'm just guessing.
I've been kicked off the ASR forum for insisting 32-bit float cannot reduce clipping in any way.
I agree. There is always a level at which an input is too hot to handle, and as far as I know there is no hardware that can fill more than 24-bit meaningfully.

Anyway, I do not fully agree with DVDdoug. Let me explain.

If you have a dynamic microphone, there is usually no amplifier in the mic like for a condensor microphone. So let us assume there are only 2 components in the chain: the pre-amp and the ADC. If a very good ADC is paired with a mediocre pre-amp, it can be the THD+N (distortion plus noise) introduced by the pre-amp is (much) louder than the THD+N of the ADC, even at the lowest gain setting. If that is the case, there is no point in turning up the gain: if the lowest gain and the highest gain result in an equal amount of noise (compared to the signal, i.e., SNR) then using the lowest gain will give the most headroom.

However, if the ADC is good and the pre-amp is good too, then usually the THD+N of the ADC will be higher than the THD+N of the pre-amp at lowest gain. If that is the case, turning up the gain will result in a higher SNR, until the point that the THD+N of the pre-amp gets louder than the THD+N of the ADC. From that point, turning up the gain doesn't improve SNR.
5
Scientific Discussion / Re: What is the pre-amp input window range for dynamic microphone voltages
Last post by Maxotics -
There are some new 32-bit floating-point interfaces.   I assume that will be getting more common.    From what I recall, this gives you a digital range of around -1000dB to +1000 dB (infinite for all practical audio purposes) and Zoom claims "no need to set gain".   But there is still analog noise and analog clipping.   You might get 20dB more usable dynamic range with a floating point interface, but I'm just guessing.
I've been kicked off the ASR forum for insisting 32-bit float cannot reduce clipping in any way.  So best I put that on another thread after this.  I wrote an essay explaining why, if you want to polish your knives  :))

Please allow me to put what you said another way.  Let me know how I'm confused/wrong.

1. When I plug a dynamic mic into an audio interface (for example) it will amplify the voltage changes near perfectly, x to y at whatever linear factor one wants
2. There will be noise, as in anything electrical, but either buried in a strong voltage or distorting a low unimportant voltage.
3. When gain is applied it amplifies all the voltage at greater factor, resulting in more noise in the final quantization (and potential clipping if the voltage is above what the ADC can handle as an input).
4. However, if I don't create a clipping voltage I will have the convenience of having a signal closer to where I'll normalize.

I've tried to test this a bit, so might not be surprised at your answer.

There is no difference in noise between a low-gain recording and a high-gain recording (assuming no clipping).  The only difference is you'll notice it immediately with the high gain recording, but will only notice it in the low-gain recording if you have to boost the levels.

Thanks again for you time and patience!  Is that correct?
6
foobar2000 mobile / Re: [Android]Start playing directly at startup
Last post by k3677 -
现在这样很好,是记录上次的播放状态。
也就是,你需要按下暂停而不是终止进程。
说实话我没想到记录上次的播放状态有啥好处,毕竟它没有记录播放位置。况且即使记录了播放位置,我也认为以pause的状态启动是更合适的。
先暂停再退出,这个操作过程比直接退出更繁琐啊。
而且这个行为跟大多数其他播放器都不一样,跟fb2k自己的桌面端也不一样,因此很难让人习惯吧。

Please provide English translation in future posts.
TOS 10. All members must post in English. Content in other languages allowed as long as full English translations are provided.

Google translate:
This is good now. It records the last playback status.
That is, you need to press pause instead of terminating the process.
To be honest, I didn't think about the benefit of recording the last playback status. After all, it doesn't record the playback position. Moreover, even if the playback position is recorded, I think it is more appropriate to start it in pause state.
Pause first and then exit. This operation process is more cumbersome than exiting directly.
Moreover, this behavior is different from most other players, and it is also different from fb2k's own desktop version, so it is difficult for people to get used to it.
8
Scientific Discussion / Re: What is the pre-amp input window range for dynamic microphone voltages
Last post by DVDdoug -
Quote
But I don't believe any pre-amp can perform accurate linear voltage multiplication with a microphone voltage/current because the current is too weak.
Linearity isn't a problem until clipping.  It's super-easy to build a highly-linear amplifier.   Non-linearity will show-up in distortion measurements and most audio amplifiers (even cheap ones) have distortion below audibility unless over-driven.

There is a type of distortion in class A/B power amplifiers called "crossover distortion" and it happens at the zero crossing where the signal goes from positive-to-negative or negative-to-positive.    Crossover distortion is worse (when measured as percent-distortion) at lower levels.   But it's normally below audibility and it's harder to hear distortion at lower levels even if it's a higher percentage of distortion.   Power amplifier output-stages are about the only place you'll find class A/B circuits.    (This is one reason some "crazy audiophiles" think class A is better.) 

At low levels, noise is the problem.   Electrical noise is "instability".   You could have a 1V signal and if the noise is 0.1V, the actual voltage will randomly fluctuate between 0.9V and 1.1V.  If it's an AC audio signal the noise and signal are summed.   If the noise is random it will be white noise.


Quote
forcing the amplifier to focus on the strong end while the weak end falls off the sensitivity of the pre-amp.
It's not "focusing" on anything.  Remember, it's just multiplication.  You can multiply 1 x 10 = 10, or 1001 x 10 = 10,010 and you're not losing any resolution with bigger numbers.   Also when you have a "wave" there is a zero-crossing twice per cycle and near the zero crossing the amplitude becomes infinitesimally small, and at that time you are amplifying very-small voltages.

Quote
So, mustn't a microphone pre-amp pick a range of voltage to amplify?
Yes.  At the low-end you are limited by noise, and by clipping at the high-end.    With very loud sounds (like a gunshot) you can sometimes use a dynamic microphone and bypass the preamp.
 
Quote
If that wasn't true we wouldn't need a gain knob in the same way there isn't a gain knob in a high end stereo system amplifier because the amplifier assumes it's receiving a voltage optimized for amplification.
There are certain "things" like the Berhinger UCA202 audio interface, or most USB turntables, that don't have a recording-level control.   They are usually calibrated for lower digital levels so they don't easily clip.  Most inexpensive USB mics don't have a control either and they also put-out low digital levels with "regular voice" but they will overload if you stick them in front of a loud guitar amp or kick-drum. 

You have to cover a super-wide range.
20dB is a factor of 10
40dB is a factor of 100
60dB is a factor of 1,000.
80 dB is a factor of 10,000

If you are getting 0.001mV from the mic and you go 60dB louder, you'll be getting 1mV.

There are some new 32-bit floating-point interfaces.   I assume that will be getting more common.    From what I recall, this gives you a digital range of around -1000dB to +1000 dB (infinite for all practical audio purposes) and Zoom claims "no need to set gain".   But there is still analog noise and analog clipping.   You might get 20dB more usable dynamic range with a floating point interface, but I'm just guessing.
9
foobar2000 mobile / Re: [Android]Start playing directly at startup
Last post by yisisixu -
现在这样很好,启动是保持上次的播放状态。
你需要按下暂停而不是终止进程。

Press pause , Don’t Terminate Process


Google translate:
This is good now. The startup is to keep the last playing state.
You need to press pause rather than kill the process.