[techtalk] can i help me

[sreeram]

ok Almut Behrens
ThankQ f your reply. as per my requirement i cannot change the hard ware.
but you are telling we can change the sampling rate frequency . is it
possible to change using software.
if so let me know how to change the sampling rate frequency.





Sreeram




----- Original Message -----
From: "Almut Behrens" <almut_behrens at yahoo.com>
To: <techtalk at linuxchix.org>
Cc: <skanumuri at email.masconit.com>
Sent: Friday, August 03, 2001 4:01 PM
Subject: Re: [techtalk] can i help me


> On Thu, Aug 02, 2001 at 03:28:35PM +0800, sreeram wrote:
> >
> > ok i am writting a program in linux 6.2 which will
> > read the file on the other side  which is having the (*.wav) data. as
every
> > data is there in the file the process is reading the data
> > very fastly that it is doing the process in very less time.
> > but i want to keep this process for some time but this is not happening?
> > infact what is happening is when it reaches the end of the file and when
> > data is not there it is just inserting silence in the other side of the
> > file.
> > i dont want to keep that silence so i think if i make this process of
> > reading with the read function is made slow the process will
> > be there for some more time
>
> Hi,
>
> I'm not exactly sure whether I understand what you are trying to
> achieve. As I read you, you intend to timestretch an audio sample
> so it plays longer than the original sample is. Is that right?
>
> Basically, having an audio file simply play longer is as easy as
> changing the playback sample frequency. This method has severe
> drawbacks, though, as it inevitably will change the pitch and other
> perceived qualities of the sound (be it music or speech).
>
> Also, it can *not* be achieved by throttling the input data stream
> (which seems to be what you are trying). Rather, you would need to
> program the chip of the soundcard to 'pull' or 'consume' individual
> values at a lower rate. (Normally, you wouldn't need to do this low
> level soundchip programming yourself -- all kinds of audio software/
> drivers already do exactly that for you.)  The proper sampling rate
> is normally stored together with the audio sample (in some header,
> depending on format). You would basically just have to modify this
> entry in the header... Also, some tools have commandline options for
> specifying the playback rate. So, for example, if you change the
> original sampling rate of 44kHz down to 22kHz, it will play twice as
> long, but pitch will change too, e.g. a tone of originally 1kHz will
> be played at 500Hz.  I guess this is not really what you want -- so I
> won't go into further details...
>
> To me, it sounds more like you want to stretch the audio sample
> while leaving other perceived qualities intact... If that's what you
> are trying to do, I'd really recommend that you quickly forget about
> the project -- sorry to disappoint you ;)
>
> Even the most clever heads of the signal processing departments and the
> electronic music hardware industry have been fighting with this problem
> for decades.
>
> At least it's definitely not as simple as inserting pauses in the data
> stream, however tiny and evenly distributed they may be...
> Timestretching audio data is more of a problem of *creating* the
> missing data basically out of nowhere, i.e. trying to extrapolate from
> the existing surrounding data. Various algorithms have been devised,
> most of them with questionable results. They typically produce clearly
> audible distortions, and applicable time stretching/compression factors
> in many cases only range from about 0.7 to 1.5.
>
> Various terms have been coined around related things: timestretching,
> time/pitch scaling, pitch shifting, etc.
> One of the modern and higher quality algorithms is MPEX, which is a
> proprietory technique developed by Prosoniq (btw, it has nothing to do
> with MPEG;)  It's based on some artificial neural network algorithm.
> The associated (commercial) software package is called TimeFactory
> (available for WinXX/Mac only).
>
> A different, equally high quality technique has been developed by
> the Roland company and is available as a musical hardware device
> called VP-9000, allowing real-time modifications of pitch, formants
> and timing.
>
> There are probably more, of course. These are just the ones I know
> of -- and I'm in no way affiliated with either Prosoniq or Roland...
>
> In case you are interested in this stuff, here are a few pointers:
>
> Overview article on algorithms -- with sound samples:
>   http://www.dspdimension.com/html/timepitch.html
>
> Lots of related links on the same site:
>   http://www.dspdimension.com
>
>
> The TimeFactory (MPEX) software:
>   http://www.prosoniq.com/html/timefactory.html
>
>   (side note: disable JavaScript before following this link -- this site
>   is one of the examples of what JavaScript should *not* be used for...)
>
> A review article:
>   www.sospubs.co.uk/sos/jul00/articles/prosoniq.htm
>
> The VP-9000 hardware solution:
>   www.roland.co.jp/worldwide/products/MI/variphrase_processor/VP-9000.html
>   www.rolandus.com/USERS/RUG/ARCHIVE/Winter_00/UT-variphrase.html
>
>
> Just in case I completely misunderstood what you are trying to do,
> I apologize in advance :)
>
> Have fun,
>
> - Almut