9:55 a.m.
Hi;
I have avoided sound technology most of my life. I have a 'tin ear' and
most music just sounds to me like somebody talking in an annoying voice.
However, recent problems with alsa mixer and/or pulseaudio has convinced
me that the time has come to spend some time learning the rudiments of
sound technology. So ...
Can someone recommend a site or manual that explains things like:
* What is the difference between alsa and pulseaudio?
* Which should I choose for the Device: field?
* What is the difference between Master, PCM, Front, Line-in, CD,
PC-Speaker etc. ?
* How is sound related to video ?
* Why are there so many files associated with producing sound?
* Etc., etc., etc.? ?
I have googled, searched Wikipedia, read the PulseAudio documentation as
well as Gnome Help. It all still leaves me befuddled -- either too
simple or two technical. Once I try to throw in the concept of various
'codecs' I am lost.
I am not asking for a personal explanation of the above questions (I
have a bunch more) but all learning suggestions gratefully received.
--
Regards Bill
Fedora 10, Gnome 2.24.2
Evo.2.24.2, Emacs 22.2.1
4:24 a.m.
New subject: Advice to an audiophobe -- If others need it.
On Sat, 2008-12-27 at 15:50 -0500, William Case wrote:
You will notice that the first 11 URL's I cite are for people
like me
who have 'tin ears' and only have a secondary interest in sound as it
relates to music. As a side note, all my life I had distinguished
music as having only three characteristics; loud or soft, high or low;
and, pleasant or annoying. It seems that I wasn't as far off the mark
as I thought I was. I will, from hereon, try to pick up on and
understand some of the more basic musical nuances I hear.
Might be worth exploring some other types of music, if you're tone deaf,
you might find something that does appeal.
As well as popular, there's a plethora of genres: Classical, jazz,
avantgarde, percussion, etc., etc.
--
[tim@localhost ~]$ uname -r
2.6.27.9-73.fc9.i686
Don't send private replies to my address, the mailbox is ignored. I
read messages from the public lists.
3:25 p.m.
New subject: Advice to an audiophobe -- If others need it.
William Case wrote:
Hi;
With special thanks to Tim and David, I have read and assembled a list
of audio explanation sites. Tim and David gave me enough context so
that when I read (or re-read) the information at these sites, things
began to make sense. I have copied the list of sites I have used for
anyone else who might be as befuddled as I was. I have tried to put
them into some logical order.
I'm a classical musician who obviously also has an interest in this
subject. If there's not a separate list for discussion of it, I'd like
to create one (I run a mailing list server). But I'm not going to be
the only member. Anyone interested?
--Russell
4:27 p.m.
William Case wrote:
David, Thank you;
If you cut and pasted these answers, I would love to know where (or what
site) you got them. If you took the time to write them off the top of
your head, I doubly thank you. You have given me enough information to
do some proper research of my own particular questions and do a small
write up for my own use.
Unfortunately the background digital audio theory comes
straight from my
electronics and communication engineering degree course (15 years ago),
but I guess became "common knowledge" to me due to interest and
continued usage, both for at home and work (where we use audio DSP black
boxes to save on needing to supply separate audio mixers, equalizers,
volume controls, and audio routing boxes to client jobs). If you are
into guitar, you might like to try a program like rakarrack. This
provides audio processing functionality similar to what those DSP boxes
can do, without the expense; but it does require a pretty decent CPU,
and an understanding of the jack audio connection kit.
While I was expanding on Tim's answer's, I found myself starting to
spout technical terms; I just backpedalled each time and tried to find
non-technical words to describe the way things are.
In terms of audio apps in fedora, my summarizing way over-simplifies
things, but hopefully it gives a good idea of the makeup of the system.
( http://www.linux.com/feature/119926
http://en.wikipedia.org/wiki/PulseAudio flowchart a few screens down
http://0pointer.de/blog/projects/guide-to-sound-apis.html for a what
sound system fits best where)
Thanks for the complement, David T.
11:57 a.m.
On Fri, 2008-12-26 at 01:04 +1100, David Timms wrote:
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.
Hi, Dave,
I work in the IC Test industry, and that is the clearest non-math
explanation I have ever read. I also teach applied DSP (fourier
analysis, time series analysis and uses of IFTs.) I have endeavored to
explain to many and varied audiences these effects, but never came up
with such clarity.
May I quote you?
Thank you,
Les Howell
8:04 a.m.
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.
12:24 p.m.
Thanks Tim;
Some of these questions were meant simply as examples of the kind of
thing I don't understand. Nonetheless, to flesh them out:
On Wed, 2008-12-24 at 08:30 +1030, Tim wrote:
On Tue, 2008-12-23 at 10:55 -0500, William Case wrote:
> * What is the difference between alsa and pulseaudio?
Major differences: Pulseaudio can produce different sounds at the same
time (e.g. your IM program can bleep at you while your music program
carries on playing music, and do so without crashes, hangs, hold-ups for
one to finish, or nasty noises), and with individual volume controls for
each (e.g. your IM bleeps subdued, while your music may be reasonably
loud).
That begs the question, when I use the volume control gui the Device:
field gives me five options:
1. HDA NVida (Alsa-mixer)
2. Analog Devices AD1986A (OSS Mixer)
3. Playback: HDA Nvida - AD198X Analog (PulseAudio Mixer)
4. Capture: Monitor of HDA NVidia - AD198x (PulseAudio Mixer)
5. Capture: HDA NVidia - AD198x Analog (PulseAudio Mixer)
HDA NVida (Alsa-mixer) seems to be the default.
1. Gives me a whole range of adjustments for different channels. (I
assume channels means different sources e.g. Master, Headphone, PCM
etc.).
2. gives me two choices and
3., 4., 5. gives me only Master.
Which should I choose and why?
Help is less than useless. There should be a man page but damned if I
can find anything helpful. man pulseaudio has techish for installing
and setting up the backend but nothing that might be useful to an audio
newbie.
In case it helps:
]$ lspci returns 00:10.1 Audio device: nVidia Corporation MCP51 High
Definition Audio (rev a2) on a ASUS M2NPV-VM motherboard; Fedora 10,
Linux kernel 2.6.27.7-134.fc10.x86_64.
If I should be using HDA NVida (Alsa-mixer), why do I have PulseAudio
options?
> * What is the difference between Master, PCM, Front,
Line-in, CD,
> PC-Speaker etc. ?
Master is the overall volume control over everything, the same control
that you're used to on your stereo system.
PCM is just the volume control for generated sounds (Waves, MP3s, etc.).
Front is the volume control for the front speakers, if you have a system
with front and rear speakers (3 - 5, or more, speaker systems). Used as
a balance control between front versus rear sound levels.
Line-in will control the volume from the analogue audio line-in in
socket (which may accept signals from something like 0.2 to 2 volts of
audio, compare that to microphone sockets, which may use something in
the range of 0.0001 to 0.010 volts, i.e. there's a large factor of
difference between line and microphone signal levels).
CD will (generally) control the volume from the (3 or 4 pin) analogue
audio cable between the CD/DVD drive and the sound card. Although it's
*possible* that systems digitally decoding the audio stream from the
data from the drive (down the IDE/SATA cable) may *also* pay attention
to that volume control, it's generally a hardware control of the line
input on the audio card.
PC speaker controls, if it's connected, the motherboard beeper volume
through the sound card. On some systems, that's a cable between the
beeper output and a sound card input, on others its handled without
additional cabling, and others it can't be done.
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.
By the way, I have a Multimedia Systems Selector frontend which gives
various perms and combs which don't make sense given the above remarks
and which 'help' says to leave alone if you are not an advanced user
which, obviously, I am not. The test beep seems to work on just about
any combination.
There is also a sound preferences gui that is set to autodetect that
seems to also test out with any setting giving a long beeeep.
It is all very confusing.
And when you do use them, it's dependent
on the card whether the nominal position for the level control is all
the way up (for simplicity's sake), or part way up (allowing you to
listen to signals that are really too low in level). And again, there's
variances as to where the partial position may be (e.g. half way, or
three quarters up). Also, what's connected to the mixer plays a role
there, whether it has a low output level that will need boosting, the
same level as the card expects, or a high level that will normally be
too much.
> * How is sound related to video ?
Sound is the sound, video is the picture... The question is too
vague
to be answerable.
Sorry, I didn't mean to be vague. Most of my sound problems over the
years seem to have been related to sound that goes with video of
different kinds - Flash being the most recent. The solution for me has
always been trial and error or to get advice from the mailing list. And
then just button push without having a clue as to what I was doing. I
wanted to read some overview that would give me a fighting chance to
diagnose problems myself.
> * Why are there so many files associated with producing
sound?
Driver-wise, there's a plethora of different sound cards, which all work
very differently, and each have differing controls available to the
user. There's a plethora of different audio codecs. For playing MIDI
files, something needs to generate the actual sounds each note will play
(that could be one file per sound, or you could make use of hardware
that does it).
That's another almost unanswerable question.
Yea tell me. It wasn't a question I expected answered individually.
But I have noticed that every time I install something involving sound I
get any number of files as dependencies. A few Fedora versions ago I
uninstalled Alsa for some forgotten reason and it ripped out almost my
entire system and I had to re-install the whole Fedora.
I thought someone might have written up a manual on sound basics to
explain all this. For example, I have looked at The Linux Document
Project for something useful but all the documents are howto's with
little emphasis on why or what. From what I read, I was left with more
questions than answers.
--
Regards Bill
Fedora 10, Gnome 2.24.2
Evo.2.24.2, Emacs 22.2.1
9:52 a.m.
David, Thank you;
If you cut and pasted these answers, I would love to know where (or what
site) you got them. If you took the time to write them off the top of
your head, I doubly thank you. You have given me enough information to
do some proper research of my own particular questions and do a small
write up for my own use.
Ps. I am top posting because there is no other place to appropriately
express my thanks.
On Fri, 2008-12-26 at 01:04 +1100, David Timms wrote:
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.
--
Regards Bill
Fedora 10, Gnome 2.24.2
Evo.2.24.2, Emacs 22.2.1
4 p.m.
On Tue, 2008-12-23 at 10:55 -0500, William Case wrote:
* What is the difference between alsa and pulseaudio?
Major differences: Pulseaudio can produce different sounds at the same
time (e.g. your IM program can bleep at you while your music program
carries on playing music, and do so without crashes, hangs, hold-ups for
one to finish, or nasty noises), and with individual volume controls for
each (e.g. your IM bleeps subdued, while your music may be reasonably
loud).
* What is the difference between Master, PCM, Front, Line-in,
CD,
PC-Speaker etc. ?
Master is the overall volume control over everything, the same control
that you're used to on your stereo system.
PCM is just the volume control for generated sounds (Waves, MP3s, etc.).
Front is the volume control for the front speakers, if you have a system
with front and rear speakers (3 - 5, or more, speaker systems). Used as
a balance control between front versus rear sound levels.
Line-in will control the volume from the analogue audio line-in in
socket (which may accept signals from something like 0.2 to 2 volts of
audio, compare that to microphone sockets, which may use something in
the range of 0.0001 to 0.010 volts, i.e. there's a large factor of
difference between line and microphone signal levels).
CD will (generally) control the volume from the (3 or 4 pin) analogue
audio cable between the CD/DVD drive and the sound card. Although it's
*possible* that systems digitally decoding the audio stream from the
data from the drive (down the IDE/SATA cable) may *also* pay attention
to that volume control, it's generally a hardware control of the line
input on the audio card.
PC speaker controls, if it's connected, the motherboard beeper volume
through the sound card. On some systems, that's a cable between the
beeper output and a sound card input, on others its handled without
additional cabling, and others it can't be done.
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. And when you do use them, it's dependent
on the card whether the nominal position for the level control is all
the way up (for simplicity's sake), or part way up (allowing you to
listen to signals that are really too low in level). And again, there's
variances as to where the partial position may be (e.g. half way, or
three quarters up). Also, what's connected to the mixer plays a role
there, whether it has a low output level that will need boosting, the
same level as the card expects, or a high level that will normally be
too much.
* How is sound related to video ?
Sound is the sound, video is the picture... The question is too vague
to be answerable.
* Why are there so many files associated with producing sound?
Driver-wise, there's a plethora of different sound cards, which all work
very differently, and each have differing controls available to the
user. There's a plethora of different audio codecs. For playing MIDI
files, something needs to generate the actual sounds each note will play
(that could be one file per sound, or you could make use of hardware
that does it).
That's another almost unanswerable question.
--
[tim@localhost ~]$ uname -r
2.6.27.7-53.fc9.i686
Don't send private replies to my address, the mailbox is ignored. I
read messages from the public lists.
2:50 p.m.
New subject: Advice to an audiophobe -- If others need it.
Hi;
With special thanks to Tim and David, I have read and assembled a list
of audio explanation sites. Tim and David gave me enough context so
that when I read (or re-read) the information at these sites, things
began to make sense. I have copied the list of sites I have used for
anyone else who might be as befuddled as I was. I have tried to put
them into some logical order.
On Tue, 2008-12-23 at 10:55 -0500, William Case wrote:
Hi;
I have avoided sound technology most of my life. I have a 'tin ear' and
most music just sounds to me like somebody talking in an annoying voice.
However, recent problems with alsa mixer and/or pulseaudio has convinced
me that the time has come to spend some time learning the rudiments of
sound technology. So ...
I have googled, searched Wikipedia, read the PulseAudio documentation as
well as Gnome Help. It all still leaves me befuddled -- either too
simple or two technical. Once I try to throw in the concept of various
'codecs' I am lost.
______________________________________
Sound (& Music) Creation and Perception
http://en.wikipedia.org/wiki/Auditory_cortex
http://en.wikipedia.org/wiki/Musical_acoustics
http://en.wikipedia.org/wiki/Place_theory
http://en.wikipedia.org/wiki/Temporal_theory
http://en.wikipedia.org/wiki/Volley_theory
http://en.wikipedia.org/wiki/Perception
http://en.wikipedia.org/wiki/Pitch_(music)
http://en.wikipedia.org/wiki/Loudness
http://en.wikipedia.org/wiki/Timbre
http://en.wikipedia.org/wiki/Harmonic
http://en.wikipedia.org/wiki/Rhythmic
___________________________________
Computer Sound - Acquiring, transferring, storing, altering and
broadcasting.
http://en.wikipedia.org/wiki/Sound_cards
http://computer.howstuffworks.com/sound-card5.htm
http://www.pctechguide.com/44SoundCards.htm
http://en.wikipedia.org/wiki/ALSA_(Linux)
http://en.wikipedia.org/wiki/PulseAudio
http://en.wikipedia.org/wiki/Codecs
http://en.wikipedia.org/wiki/Open_source_codecs_and_containers
http://en.wikipedia.org/wiki/List_of_codecs
http://en.wikipedia.org/wiki/Ogg
http://en.wikipedia.org/wiki/Vorbis
http://en.wikipedia.org/wiki/GStreamer
http://en.wikipedia.org/wiki/Totem_(media_player)
http://en.wikipedia.org/wiki/Firewire
http://en.wikipedia.org/wiki/List_of_Linux_audio_software
Musical Instrument Digital Interface
http://en.wikipedia.org/wiki/MIDI
On Wikipedia, I recommend checking "See also", "References" and
"External Links" at the bottom of each page to find further and deeper
explanations.
The main breakthrough for me was being shown that ALSA was a driver for
the sound card (chip) hardware, while PulseAudio was a server program,
in much the same way that 'ati' or 'nv' are video card drivers and X
windows is a window graphics server.
All other sound programs, codecs etc. are, at heart, just libraries used
by either ALSA or PulseAudio.
It is best to take a look at http://en.wikipedia.org/wiki/PulseAudio
flowchart a few screens down as recommended by David Timms just to get a
visual road map before starting in reading.
You will notice that the first 11 URL's I cite are for people like me
who have 'tin ears' and only have a secondary interest in sound as it
relates to music. As a side note, all my life I had distinguished music
as having only three characteristics; loud or soft, high or low; and,
pleasant or annoying. It seems that I wasn't as far off the mark as I
thought I was. I will, from hereon, try to pick up on and understand
some of the more basic musical nuances I hear.
Thanks all.
--
Regards Bill
Fedora 10, Gnome 2.24.2
Evo.2.24.2, Emacs 22.2.1
5650
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