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Author Topic: Wav files  (Read 3964 times)

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Damascus

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17 November 2003, 19:41:29
Is there a size limit for .wav files in OrbiterSounds? I was trying to create some background music
and converted three tracks off a CD into .wav files. Using the Sequential Editor, I thought I set
them up properly, but the second track played over the first after a short while. The next question
I have is: Is the timing column in seconds until the next file in the sequence starts? If so, I'm
stymied, because I made the number larger than the track time.

After I couldn't get that to work, I combined the three tracks into one, 128mb, track. When I
switch to the frequency, there is an obvious pause in Orbiter and some hard drive action, which I
imagine is the .wav file being loaded. But it doesn't play. A lack of memory perhaps? I have 512mb
of RAM, but if the whole file needed to laod, I can see the problem.

Which brings me back to question 1, how do you set the timing?


Offline DanSteph

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Reply #1 - 17 November 2003, 19:59:48
Gee, 128 Mb in one file ?   :fool: 8o

First the pause you see is your computer is the 128mb
loading in ram as you supposed.

I didn't even tried loading a sound as large as that
but I would not be surprised if it cause a problem.  

What you can do is to split your songs beetween
frequency say song 1-2-3 on one frequency then
4-5-6 on one other etc etc.
You should make some test but I think that 30mo
in one wav *should* play without flaw. (perhaps
more than that, make test)

Second think you might do is to reduce if your wav are 44'100hz
to at least 22'050 this will divide your music's size by two
I recall that humain hear often don't even hear 22000hz frequency
so 44'000 is *useless*

Third things to do is perhaps to load winamp in background with your music
in mp3 ? (or OGG the format that unlike MP3 is really free, open source and better
quality than mp3 for the same size) (winamp can read *.ogg without problem)

Dan


Damascus

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Reply #2 - 18 November 2003, 05:29:07
Yep, 128mb. It's about 20 minutes of Mozart. Each track plays fine by itself so I figured the huge
file didn't fit into memory. Is the the Timing section of the Sequential Editor in seconds?


Damascus

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Reply #3 - 18 November 2003, 09:07:38
Knokced the file from 44100 to 22050. It's the same size.


Offline DanSteph

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Reply #4 - 18 November 2003, 19:36:47
Damascus wrote:
>is the the Timing section of the Sequential Editor in seconds?

Yes

> Knokced the file from 44100 to 22050. It's the same size.

That would be the first time in world that reducing the frequency doesn't
reduce the size :)
Check closer your manipulation there should be an error somewhere.

What software do you use ? Maybe I can help ?
(soundforge,cubase, cool edit  etc etc)


Dan


Damascus

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Reply #5 - 20 November 2003, 00:20:43
Just played (for the first time) the 22050 file and it's at half-speed, which may explain a lot. It's
obviously a lot easier to take each (smaller) track and stick it in the Sequential Editor. Is the Timing
column when the track will start? For example, if I have three tracks and I set the Timing column to
0, 60, 180, does Track 1 start immediately, Tracks 2 starts a minute later and Track 3 three
minutes after Track 1?

I am using Audacity, a free program since I don't do anything much with sound files. Thanks for the
help, though. I can't imagine Orbiter without the sounds.


Offline DanSteph

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Reply #6 - 20 November 2003, 00:35:39
Damascus wrote:
> Just played (for the first time) the 22050 file and it's at
> half-speed, which may explain a lot.

Yes it explain, in fact you just changed the speed of playing
without changing anything in the Data.

So you play at 22050hz a sample that contain 44'100hz of data
=it play at half the speed but you still have the same size.

You need to search something like "resample" or "change format"
I don't think it should be in "timing" option.

I don't know audacity but I can help with another soft
Check mail I'll write to you about that.

Cya

Dan


Offline bgreman

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Reply #7 - 20 November 2003, 01:55:30
Um, Dan, just a word of advice:  You're right, humans can't hear much above 20000 Hz.

But that isn't what the Hz measurements on a .wav file are for.  They measure the sample rate,
not the pitch frequency of the sound IN the file.  They measure the number of sound-bits per
second that the sound file will play on.  Thus, a 44000 Hz file will have twice as many sound bits
as a 22000 Hz file.  If you "looked" at the sound on a close scale, it will be 'smoother'.   Thus, the
Hz measurements are a measurement of the smoothness quality of the .wav file.

Make sense?


CRASHBANGRUMBLEBUMPHISS

Houston...we've flushed our toilet.

Offline DanSteph

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Reply #8 - 20 November 2003, 02:45:34
bgreman wrote:
> Hz measurements are a measurement of the smoothness quality of
> the .wav file.
>
> Make sense?


Mhhh :) you speak to a former musician (guitar/piano) that
have also a federal certificate of high school in Tv/radio/hifi electronic
(well it's dated for 20 years also but I bet that the sound and
the shape of electron didn't changed since
:music:

I understand what you mean and your right *in a certain manner* but
it's easier to consider that a 22'0050 hz wav cut frequency over 22'050hz...
In brief they are "still here" but some extract only as a text that you
cut half letter.

Anyway most people won't hear a difference beetween 44khz and 22khz.
sample that's the point.

Dan


Offline CarlosTheTackle

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Reply #9 - 20 November 2003, 05:39:50
Actually most people would quite easily hear the difference, although probably not with lo-fi radio
voices.

A fundamental principle of digital audio is that the a given sample rate can only accurately code for
audio frequencies up to 1/2 the sample. This is the Nyquist frequency. That means that 44.1kHz can
have audio material up to 22050Hz, which is why it is a handy sample rate for human hearing.

Also, it is important to make sure a low-pass filter is applied to a signal before down-sampling. In
other words, if re-sampling a 44.1 sample to 22, you need to make sure that it has had all audio
frequencies over 11kHz removed before it is re-sampled. Otherwise you can get a nasty artifact
called 'aliasing'. If anyone's interested, see http://www.earlevel.com/Digital%20Audio/Aliasing.html
for a better explanation.


Offline DanSteph

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Reply #10 - 20 November 2003, 20:09:11
I should write "won't notice the *bad quality* of the 22'050hz sample unless you make
them hear both one after the other" playing with stereo help greatly to enhance a
22'050 sample also.

Most humain don't pay to much attention to sound quality...  (otherwise we would not
see all those "hifi" with incorporated reverb)  And the brain is wonderful in a sense that
unless the sound is really bad it can compensate for what's missing.

About aliasing , most soft even old have anti aliasing built in or as an option
when you resample.

Dan

« Last Edit: 20 November 2003, 20:09:11 by DanSteph »