Playback settings
A part of the interest for foobar comes from different technical decoding and sending data options. If you want to have a good sound, take care to the following settings which will influence the listening. Notice that two parts are important:
Secondary settings
This first part of the playback section allows you to do some minor settings like
- the length of the pre-buffering requisit (buffer files up to), selected according to the user wanted size. It hasn't any impact on the audio quality but could solve some cracks dued to "accessing to the file" problems which can occure if your hard drive is a bit ousted. It refers only to files with a smaller size than the chosen threshold.
- the Playback thread priority defines the level of priority given to foobar by the system when this software tries to send datas to your audio card.
- a low value could create some interference
- choose max. by default, reduce the priority if you encounter some problems only.
This setting is independant from the diskwriter proposed thread priority
- the "Show clipping warning" box might be ticked by clipping trackers (it will show them if their settings are good ones). Otherway, untick it to avoid some pop up from the console while playing.
- the "volume control" cursor. Some users are surprised to have this field avaible only from this panel. But as the sound is setted with windows your audio card and the player, the level value is quite unsignificant. With replaygain and the preamp, the software may smooth noise level problems and the need to change it's value. Changing it should be done only for exceptional circumstancies like phone calls or the hour (so as not to disturb your neighbours). And there are usually potentiometers on your hardware.
It's unactivated by default. To activate it, go to the DSP manager.
- Note: the value in dB is calculated with this formula:
dB gain=20log(I/IO) where I=final value and I0 the reference
It's always a negative value. -3db corresponds to dividing the output level by sqrt(2)
this is a table for the correspondance between the percentage value and the dB value (after -60dB, it's nearly unhearable even if your hardware is put to max.):
|
in %
|
in dB
|
in %
|
in dB
|
|
100
|
0
|
30
|
-10.4
|
|
90
|
-0.9
|
20
|
-14
|
|
80
|
-1.9
|
10
|
-20
|
|
70
|
-3.1
|
1
|
-40
|
|
60
|
-4.4
|
0.1
|
-60
|
|
50
|
-6
|
0.01
|
-80
|
|
40
|
-8
|
0.001
|
-100
|
Bit depth
- Bit depth, the audionumeric resolution creates a lot of phantasms. For the moment, most audio files are encoded in 16 bit (audio CD for example) and most audio cards support only 16 bit informations. And it's hard to notice that this depth is in default. As 24 bit audio cards have appeared, foobar can fully support their bit depth and even the next generations of audio cards. The improvement could be noticeable for those cards as foobar manages padding, which consists in sending "dead" values if the audio card depth is higher than the resolution of the source (as does the Terratec DMX6fire 24/96) (it does so only for the "extra bit" values !!).
Which depth should you use?
It highly depends on your audio card. The best is the maximum depth of your audio card. Just take care to check if the depth given from the producer is the same as the real value. For example, the creative audigy only supports 24bit data for transit, not for working.
Show all option should be used with caution as it hasn't been proved to have a real impact on the listening.
Use 16 bit mode by default, change only if you notice hearable problems and know that your sound card can support the mode selected.
If you have an audio card which works in 24 bit mode, you can choose by yourself which depth is better. (You can ask a frien to switch between two modes and keep the one you found best)
- Dithering and noise shaping. foobar works natively into 64 bit float mode, as a consequence, your card will have to delete some information calculated by the software. The more your data is sampled, the more it will be accurate. You can convince yourself with increasing from 2 bit to 16 bit while playing, the breathing (I'm not sure of the word) while change from real blizzard to nothing.
There are several ways to reduce the information:
- trimming, simply cuts all that is above the bit depth chosen, not the best as it is quite primary
- dithering, the final word is recalculated, taking in count the informations given by the higher bits. It's particularly important for low level parts of track were trimming could be a real disaster. Dithering is a way to resample datas in a clever way, as it includes a bit more information than your hardware could normally support. On the other way, background noises could be more hearable (especially for sampling to low resolution). Generally, dithering is better than trimming, especially at 16 bit, as the additional breathe is always at a very low level, generally out from the hearable zone and totally inconsequential compared to the noise of your computer for example. Moreover, their are several ways to put the breathing away from the perception field. If there are few noise shaping technics, their implemention are various: each software producer has its proper algorithm, more or less efficient from one sofware to another, depending on the listener's taste (and their hardware). Combined to noise shaping, dithering allows the user to have a resolution above it's hardware capacities. without having hearble enhancemnt of the background noise. Foobar offers three mode of noise shaping to use with dithering:
The strong mode is recommended.
The hearable improvement dued to dithering decreases as the bit depth used increased. As a consequence, people using 24 bit depth hardware could switch off this option if they want to preserve their system resources. (dithering at that depth couldn't be hearable even with the best hardware as your computer produces too much noise to hear the enhancement (and even without that..))
Output
Output:
There are several outputs to carry information from the sofware to your audio card. These ways correspond to different generations of softwares, waveout is the oldest, direcsound the most used, kernel streaming the most modern one, and ASIO which is apart.
- Waveout: being old doesn't mean being bad, as it the most compatible with every kind of audio card drivers, if you're experiencing some trouble with other output modes, choose this one. It might stop. There is no special options except setting the length of the buffer.
- Direct sound: widespread, it have some problems with some audi card, could crash foobar (but it's rare), or even system crash (very very rare!!). You can setup the length of the buffer and allow hardware mixing (activate or not with the components installed, or problems with foobar)
- Direct sound V 2.0: allows fading options at certain moments. (fading is different from crossfading, it does'nt mix two tracks, only decreases the audio level of the track). If crashes repetead after activating this output, especially while changing of playlist, think to change your output.
- Kernel streaming: it's a transit channel so as to bypass the system mixing. The sound comes from the software directly to the audio card, but on the other hand, it has several problems of compatibility, usually linked to the audio card and its drivers. It may cause foobar crashes..
- kernel streaming needs a perfect relevance between the sampling of audio data and the working pace of the audio card (16bit
16 bit, 32 bit 32 bit) in case of irrelevancy an error message will appear to warn the user.
- ASIO, it's the same as kernel streaming but is older, use it as you do with kernel streaming
which one should you use?
- from theorical point, kernel streaming and ASIO technics might prove to be the best
- if you do not want to have stability problems for foobar, prefer waveout
- if you need fading, use Directsound V2.0
In case of repeated crashes think to the output settings and the parameters chosen.
Q: I have no sound at all for my mp3 while the song is played perfectly in other readers, What is the problem?
A.