| Odesílatel: [EMAIL PROTECTED]
| I grabbed Sprenger's smsPitchScale.cp from
http://www.dspdimension.com/
| and wrapped a bit of support code around it. This program will change
the
| pitch of an audio file - this means you can play it back so that talking
is
| twice as fast but *at the
On 14-Sep-2000 [EMAIL PROTECTED] wrote:
Does anyone know a good technic/routine to time-stretch a buffer of audio
data
Also on this note:
I want to replay audio of talking, but I want it sped up without too much of a
change in pitch. I'm sure someone must have worked through this problem
To answer my own post,
I grabbed Sprenger's smsPitchScale.cp from http://www.dspdimension.com/
and wrapped a bit of support code around it. This program will change the
pitch of an audio file - this means you can play it back so that talking is
twice as fast but *at the same pitch* i.e.
Howdy Mike,
Wow, great pointer. The DSP Dimension page had one of the most concise and
yet clear discussions of pitch scaling I've seen. (And it's nice to know
that I didn't need to hedge so much in my original response - my memory was
pretty much spot on...) I'm going to have to read through
Howdy Steve,
Does anyone know a good technic/routine to time-stretch a
buffer of audio data
(mono channel) ?
Assuming that you want to keep the pitch the same, phase vocoding comes to
mind. This is an old speech processing trick (thus the vocoder) for
shifting pitch without changing speed
Hi!
Does anyone know a good technic/routine to time-stretch a buffer of
audio data (mono channel) ?
In the case you want to replay the audio data at a different sampling rate
WITH changing the duration of the sound accordingly to the ratio of
original sampling rate and new sampling rate you
will be
different...
So what I need would be something to change the pitch and not the sampling
freq (or better the duration of the sample and no pitch difference)
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 14, 2000 4:33 PM
Subject: RE: [MP3 ENCODER] Time
From: "Steve Lhomme" [EMAIL PROTECTED]
Resampling just add/delete samples from the original
If you mean that literally, that's a very bad way to do it. :) Take a look
at SoX, which provides several resampling algorithms with different
tradeoffs. I think the resampler in LAME is roughly
On Thu, Sep 14, 2000 at 09:48:04PM +0200, Markus Fick wrote:
In the case you want to change the sampling frequency WITHOUT changing the
duration of the sound (or changing the sound duration without following the
ratio of original to replay sampling rate) the following link is a good
point
Looking at the (limited) docs for the demo Fraunhofer encoder (V3.1
Demo (build Sep 23 1998)), one of the options is to enable/disable
"Soft time-domain filtering" which they say uses "a high-quality time
domain filter instead of fast MDCT".
They must be referring to the psycho acoustic model,
On Tue, 19 Oct 1999, Mathew Hendry wrote:
Could someone examine this ?
http://www.isoternet.org/~tominaga/lame-beta/lame-1017-4.tar.bz2
oh, and the "time/estimated" display is somewhat messed up:
Encoding as 44.1kHz 128 kbps j-stereo MPEG1 LayerIII file
Frame |
..just noticed something weird with the timestatus that seems to occur when
encoding MPEG2-layer3...
It seems to report being twice as fast as it really is, this also seems to
only occur on stereo samples, so might it be that it adds both channels into
the timeline instead of treating them as
On Sat, 9 Oct 1999, Mathew Hendry wrote:
It seems to be because totalframes is calculated incorrectly for MPEG 2
streams. You'll notice that the "% complete" indicator is wrong too.
this was because LAME was computing 'totalframes' before figuring
out if it was doing mpeg1 or mpeg2, and so
Anyone knows how much time is spent in each function?
Gabriel Bouvigne - France
[EMAIL PROTECTED]
icq: 12138873
MP3' Tech: www.mp3tech.org
--
MP3 ENCODER mailing list
14 matches
Mail list logo
