Advice to an audiophobe ??

[David Timms]

William Case wrote:
> On Wed, 2008-12-24 at 08:30 +1030, Tim wrote:
>> On Tue, 2008-12-23 at 10:55 -0500, William Case wrote:
...
> 1. Gives me a whole range of adjustments for different channels.  (I
> assume channels means different sources  e.g. Master, Headphone, PCM etc.).
Just to clarify, a source device generates an audio signal( line in, PCM 
(pulse code modulation = wav uncompressed audio), that might then get 
processed (eg volume control, master, headphone), and then sent to a 
destination (often an output device like a speaker connector or 
headphone jack).

Because a typical soundcard has an internal hardware mixer, it can 
usually mix together various inputs (sources) like CD input, mic input 
and recorded audio signals, and produce a single output signal (mostly 
in stereo=2 channels). When mixing together externally received signals, 
no main CPU processing power is used, unless you are trying to record to 
hard disk etc.

> 2. gives me two choices and 
> 3., 4., 5. gives me only Master.  
> Which should I choose and why?
capture means recording - usually from a line in (eg from a vcr or mp3 
player etc), or from a microphone. So exclude those from your choice.

> If I should be using HDA NVida (Alsa-mixer), why do I have PulseAudio
> options?
Consider pulse audio to be a real-time digital mixer and volume control, 
where the audio calculations are performed inside your main CPU. In the 
default setup, once pulseaudio has done it's processing, it passes the 
result to the alsa driver which outputs the audio data to the soundcard. 
The soundcard turns the digital audio data into analog audio signals for 
  use with amplifier, speakers, or headphones.

Pulseaudio also has enhanced capabilities like remembering that when you 
playback with xmms that you like to output via your amplifiers and 
speakers, but when you are viewing a flash video, to playback into your 
headphones instead, at a different level. Another capability let's you 
choose the destination playback device while the material is actually 
being played. A third capability let's the output go to an audio device 
on another machine. Obviously, this is a bit trickier to set up.

...
>> These individual mixer input controls should normally be left off if you
>> never use them, as they can each introduce noise (hiss, beeps and
>> burbles, etc.) to the system.  
> I will turn them off except for Master and Front.  I will experiment
> with PC Speaker.  Of course these are only available to me if I use the
> default alsa mixer setting.
It's not a one or other setting, both parts will still be involved; 
pulseaudio will process, mix, and attenuate sound signals, whereas alsa 
will drive the physical hardware. The setting you are seeing lets you 
decide whether to control the physical driver volume levels or the 
software generated pulseaudio volume controls. If you mute or turn the 
alsa master way down, it wont matter how high you turn the pulseaudio 
mixer, since the alsamixer comes after the pulseaudio one in the audio 
chain. (also true for the reverse).

If you play back a loud audio file, and turn both the pulseaudio source 
  and master up full. Then change to the alsa setting. You can then use 
the also setting to set up an absolute maximum level that you would want 
to hear, by adjusting the master. Then you could go back to the 
pulseaudio setting to adjust the playback to a comfortable setting, and 
from then on only use the pulseaudio setting.

...
>>>       * How is sound related to video ?
> 
>> Sound is the sound, video is the picture...  The question is too vague
>> to be answerable.
In digital format, sound and vision are both represented with digital 
1's and 0's. With all video and audio file types, there is a packing 
together of the audio and video information into the one file. The 
multiplexed file provides information about when to playback each frame 
of video in relation to the audio in the file. For example, an mpeg2 
(dvd) file might have two frames of video, then 2 of audio, then 1 of 
video, two audio in an order to achieve a consistent throughput of audio 
and video data.

>>>       * Why are there so many files associated with producing sound?
In digital audio, the most basic file type is waveform (.wav), where 
each momentary value of audio is stored, on a 1 for 1 basis. Experiments 
and calculations can show us that for something we store as quality 
musical recording we need to sample that momentary value at 44kHz (times 
per second) or higher so as not to disrupt our digital recording with 
audio aliases. Since we also seem to enjoy the spatial enhancement 
produced by stereo or more channels, the file needs to store both left 
and right information. Finally, we found that if we only store the 
digital value using a small no of bits per sample, when played back we 
hear a harsh, chunky sound, rather than the CD like quality of using 16 
(or more) bits per sample. The catch with all that is it takes up a lot 
space.

To solve space issues (less a problem now that storage space costs a lot 
less), compression schemes were developed. Most take advantage of 
reducing the number of channels eg to mono, reducing the sampling rate, 
or the number quantizing levels (bits/sample); but this is done in 
context of the type of audio being compressed - eg human voices are 
typically of lower frequency, and can sampled at a slower rate, and with 
less levels.

The biggest jump in compression was with psychoacoustic modelling, where 
it was found that in a complex sound, a listener does not notice that 
certain frequency (pitch) sounds become inaudible (or masked) by other 
sounds.

The reason there is so many formats, is because developers were 
essentially competing to produce more highly compressed audio files, 
without noticeable change in quality, when using a certain type of 
audio, over a certain communication medium. Eg: when the fastest home 
internet connections were slow modems, compression made it possible to 
transmit voice signals over your internet connection. If you tried to 
transmit music of higher quality that voice, you would have large audio 
distortions that made it difficult to hear the original material.

You might like to play with the audio editor program audacity (perhaps 
from rpmfusion if you want to be able to import and save in certain 
compressed formats (mp3)). It shows you a graphical representation of 
the audio file, and eg lets you choose a zoom, start and stop position, 
and just play back small parts of a file, so that you can work out what 
the sound "looks" like to a computer.

Hope that helps a bit more ;-)
DaveT.