Reply by Green Xenon [Radium]●September 12, 20072007-09-12
Ron N. wrote:
> Pitch shifting
> downwards either throws away time domain snippets or
> frequency bins, or increases the total duration of the audio.
The frequencies are downshifted but the shapes of the audio signals
remain similar.
Reply by Ron N.●September 12, 20072007-09-12
On Sep 11, 5:06 pm, "Green Xenon [Radium]" <gluceg...@excite.com>
wrote:
> "Time and pitch processing: shift pitch without changing tempo - and
> never introduce audio artifacts."
>
> If the pitch of the audio can be decreased this way, then
The problem is that in the general case it can't. Pitch shifting
downwards either throws away time domain snippets or
frequency bins, or increases the total duration of the audio.
Sometimes there is no audible information in the snippets or
bins which the processing throws away, and you can compress
the bits down closer to their information content (like an mp3
coder tries to). Sometimes there is audible information,
and too much compression will sound bad.
IMHO. YMMV.
Reply by Jerry Avins●September 12, 20072007-09-12
cincydsp@gmail.com wrote:
> -snip-
>
>> Has this technique ever been tried in the past? If so, was it as
>> efficient as I think it would be?
>>
>> Thanks,
>>
>> Radium
>
> As always, what's important is the total bandwidth of the signal.
> Shifting things around in frequency doesn't buy you anything; your
> minimum sampling frequency is still determined by the total bandwidth
> of the signal, which you can't change without distorting it in some
> way. Basically, all you're describing is shifting some bandpass signal
> down to (almost) baseband, which doesn't do anything for you with
> respect to compressibility.
No, no, the trap he laid for himself is more seductive than that. He's
contemplating pitch shifting, not frequency shifting. As conservation of
energy rules out perpetual-motion devices, just so does conservation of
information rule out his compression scheme. I won't write more about
why until he's had time to come up with a coherent explanation.
Jerry
--
Engineering is the art of making what you want from things you can get.
¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
Reply by ●September 12, 20072007-09-12
-snip-
> Has this technique ever been tried in the past? If so, was it as
> efficient as I think it would be?
>
> Thanks,
>
> Radium
As always, what's important is the total bandwidth of the signal.
Shifting things around in frequency doesn't buy you anything; your
minimum sampling frequency is still determined by the total bandwidth
of the signal, which you can't change without distorting it in some
way. Basically, all you're describing is shifting some bandpass signal
down to (almost) baseband, which doesn't do anything for you with
respect to compressibility.
Jason
Reply by Jerry Avins●September 12, 20072007-09-12
Green Xenon [Radium] wrote:
> Hi:
>
> Decreasing the pitch of the audio in an audio linear-PCM [wave] file
> means the sample-rate of the wave file can be decreased without causing
> aliasing. If the bit-resolution and # of channels [1 in mono, 2 in
> stereo] of the file are kept constant, then decreasing the sample-rate
> will decrease the file size. Adobe Audition allows the alteration of
> pitch without changing speed.
>
> http://www.adobe.com/products/audition/overview2.html#kmhead3
>
> "Time and pitch processing: shift pitch without changing tempo - and
> never introduce audio artifacts."
>
> If the pitch of the audio can be decreased this way, then sample-rate
> can also be decreased until it is just 2.5x the maximum frequency of the
> audio signal in the file. Mathematically, only 2x the max frequency is
> necessary, however, due to physical factors, its best to use at least
> 2.5x the max frequency.
>
> Suppose it is desired to decrease the file size, keep the good audio
> quality, and not use any compression. In this case, the pitch of the
> audio can be decreased sufficiently so the sample-rate can also be
> decreased sufficiently to store this file in devices with not much space
> or transfer on the internet on low-bandwidths. Instructions regarding
> how the file was processed and what the audio signals and sample-rate
> were like before the processing can be stored with the file. Upon
> reading the file, these instructions tell the software to increase the
> sample-rate and pitch of the audio file back to what is was prior to the
> processing. Then the audio can be played back w/out any artifacts.
>
> For best results, the pitch would be decreased all the way until the
> lowest-frequency signal can finish its cycle in the appropriate amount
> of time � e.g. if a song is 5 minutes long, the lowest frequency signal
> is downshifted until it has a frequency of 1 cycle every 5 minutes. This
> ensures that the information is not lost or cut-off. After this, the
> software would assign an appropriate sample rate for the file. If the
> highest frequency signal is 1,000 Hz, then the sample rate would be
> 2,500 Hz.
>
> Has this technique ever been tried in the past? If so, was it as
> efficient as I think it would be?
The scheme won't work. I leave it to you to think for once. Silly
questions merely advertise your ignorance.
Jerry
--
Engineering is the art of making what you want from things you can get.
¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
Reply by Jerry Avins●September 11, 20072007-09-11
Green Xenon [Radium] wrote:
> Hi:
>
> Decreasing the pitch of the audio in an audio linear-PCM [wave] file
> means the sample-rate of the wave file can be decreased without causing
> aliasing. If the bit-resolution and # of channels [1 in mono, 2 in
> stereo] of the file are kept constant, then decreasing the sample-rate
> will decrease the file size. Adobe Audition allows the alteration of
> pitch without changing speed.
>
> http://www.adobe.com/products/audition/overview2.html#kmhead3
>
> "Time and pitch processing: shift pitch without changing tempo - and
> never introduce audio artifacts."
>
> If the pitch of the audio can be decreased this way, then sample-rate
> can also be decreased until it is just 2.5x the maximum frequency of the
> audio signal in the file. Mathematically, only 2x the max frequency is
> necessary, however, due to physical factors, its best to use at least
> 2.5x the max frequency.
>
> Suppose it is desired to decrease the file size, keep the good audio
> quality, and not use any compression. In this case, the pitch of the
> audio can be decreased sufficiently so the sample-rate can also be
> decreased sufficiently to store this file in devices with not much space
> or transfer on the internet on low-bandwidths. Instructions regarding
> how the file was processed and what the audio signals and sample-rate
> were like before the processing can be stored with the file. Upon
> reading the file, these instructions tell the software to increase the
> sample-rate and pitch of the audio file back to what is was prior to the
> processing. Then the audio can be played back w/out any artifacts.
>
> For best results, the pitch would be decreased all the way until the
> lowest-frequency signal can finish its cycle in the appropriate amount
> of time � e.g. if a song is 5 minutes long, the lowest frequency signal
> is downshifted until it has a frequency of 1 cycle every 5 minutes. This
> ensures that the information is not lost or cut-off. After this, the
> software would assign an appropriate sample rate for the file. If the
> highest frequency signal is 1,000 Hz, then the sample rate would be
> 2,500 Hz.
>
> Has this technique ever been tried in the past? If so, was it as
> efficient as I think it would be?
>
>
> Thanks,
>
> Radium
--
Engineering is the art of making what you want from things you can get.
¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
Reply by Green Xenon [Radium]●September 11, 20072007-09-11
Hi:
Decreasing the pitch of the audio in an audio linear-PCM [wave] file
means the sample-rate of the wave file can be decreased without causing
aliasing. If the bit-resolution and # of channels [1 in mono, 2 in
stereo] of the file are kept constant, then decreasing the sample-rate
will decrease the file size. Adobe Audition allows the alteration of
pitch without changing speed.
http://www.adobe.com/products/audition/overview2.html#kmhead3
"Time and pitch processing: shift pitch without changing tempo - and
never introduce audio artifacts."
If the pitch of the audio can be decreased this way, then sample-rate
can also be decreased until it is just 2.5x the maximum frequency of the
audio signal in the file. Mathematically, only 2x the max frequency is
necessary, however, due to physical factors, its best to use at least
2.5x the max frequency.
Suppose it is desired to decrease the file size, keep the good audio
quality, and not use any compression. In this case, the pitch of the
audio can be decreased sufficiently so the sample-rate can also be
decreased sufficiently to store this file in devices with not much space
or transfer on the internet on low-bandwidths. Instructions regarding
how the file was processed and what the audio signals and sample-rate
were like before the processing can be stored with the file. Upon
reading the file, these instructions tell the software to increase the
sample-rate and pitch of the audio file back to what is was prior to the
processing. Then the audio can be played back w/out any artifacts.
For best results, the pitch would be decreased all the way until the
lowest-frequency signal can finish its cycle in the appropriate amount
of time � e.g. if a song is 5 minutes long, the lowest frequency signal
is downshifted until it has a frequency of 1 cycle every 5 minutes. This
ensures that the information is not lost or cut-off. After this, the
software would assign an appropriate sample rate for the file. If the
highest frequency signal is 1,000 Hz, then the sample rate would be
2,500 Hz.
Has this technique ever been tried in the past? If so, was it as
efficient as I think it would be?
Thanks,
Radium