I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is there a
program that can clean up the file (IE, remove the hiss and crackle and or
improve overall sound quality)?

Thanks,
DV

I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is there a
program that can clean up the file (IE, remove the hiss and crackle and or
improve overall sound quality)?

Unfortunately, it's likely that you did some very difficult-to-repair
damage to the audio quality. By recording at 8 bits of resolution,
and not having a proper "dither" in the process, you will have
generated a great deal of quantization noise in the signal. This
noise (the crackling and hiss that you hear) is, mathematically, quite
a mess... some of it may be harmonically related to the original
signal, some of it will be intermodulation between the various parts
of the signal, and much of it is probably broadband in nature.

You're never going to be able to process this so that it sounds
anywhere nearly as good as a 16-bit capture would have sounded. All
of the low-level detail in the signal has been lost due to truncation.
The best you're going to be able to do is make it somewhat less
awful-sounding.

I would guess that you ought to be able to use SoundForge, CoolEdit
(if you have a copy), DART, DC-ART, or any of a number of similar
sound tools to clean up the files somewhat. You'll probably need to
do it in several steps.

What I'd probably try, myself, would be the following:

- Convert the file to 16-bit format at the same sampling rate, to give
yourself some processing headroom.

- There may be a "crackle" filter available - if so, experiment with
it and see if it helps.

- Apply a low-pass filter, to get rid of the higher-frequency
harmonic and intermodulation distortion and hiss. You'll
necessarily be sacrificing some of the treble in the signal if you
do this. Play around with the filtering and equalization, to get
the best tradeoff between noise reduction and loss-of-treble.

- Apply a dynamic noise-reduction filter - one which adapts in real
time to the amplitude of the signal. Some of these are low-pass
filters with automatic adjustment of the filter "knee" frequency
and slope. Others are multi-band filter banks, where each bank
will pass its specified frequency range only when there's more than
a certain amount of signal present.


As a quickie hack, you could try converting one of these files to
16-bit format, and then running it through an MP3 encoder at a
relatively low encoding bit rate. This _might_ cause some of the hiss
and noise to be discarded as "not musically or sonically worth
keeping", and the more musical frequency bands to be retained.

--
Dave Platt AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is there a
program that can clean up the file (IE, remove the hiss and crackle and or
improve overall sound quality)?

Unfortunately, it's likely that you did some very difficult-to-repair
damage to the audio quality. By recording at 8 bits of resolution,
and not having a proper "dither" in the process, you will have
generated a great deal of quantization noise in the signal. This
noise (the crackling and hiss that you hear) is, mathematically, quite
a mess... some of it may be harmonically related to the original
signal, some of it will be intermodulation between the various parts
of the signal, and much of it is probably broadband in nature.

You're never going to be able to process this so that it sounds
anywhere nearly as good as a 16-bit capture would have sounded. All
of the low-level detail in the signal has been lost due to truncation.
The best you're going to be able to do is make it somewhat less
awful-sounding.

I would guess that you ought to be able to use SoundForge, CoolEdit
(if you have a copy), DART, DC-ART, or any of a number of similar
sound tools to clean up the files somewhat. You'll probably need to
do it in several steps.

What I'd probably try, myself, would be the following:

- Convert the file to 16-bit format at the same sampling rate, to give
yourself some processing headroom.

- There may be a "crackle" filter available - if so, experiment with
it and see if it helps.

- Apply a low-pass filter, to get rid of the higher-frequency
harmonic and intermodulation distortion and hiss. You'll
necessarily be sacrificing some of the treble in the signal if you
do this. Play around with the filtering and equalization, to get
the best tradeoff between noise reduction and loss-of-treble.

- Apply a dynamic noise-reduction filter - one which adapts in real
time to the amplitude of the signal. Some of these are low-pass
filters with automatic adjustment of the filter "knee" frequency
and slope. Others are multi-band filter banks, where each bank
will pass its specified frequency range only when there's more than
a certain amount of signal present.


As a quickie hack, you could try converting one of these files to
16-bit format, and then running it through an MP3 encoder at a
relatively low encoding bit rate. This _might_ cause some of the hiss
and noise to be discarded as "not musically or sonically worth
keeping", and the more musical frequency bands to be retained.

--
Dave Platt AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is there a
program that can clean up the file (IE, remove the hiss and crackle and or
improve overall sound quality)?

Unfortunately, it's likely that you did some very difficult-to-repair
damage to the audio quality. By recording at 8 bits of resolution,
and not having a proper "dither" in the process, you will have
generated a great deal of quantization noise in the signal. This
noise (the crackling and hiss that you hear) is, mathematically, quite
a mess... some of it may be harmonically related to the original
signal, some of it will be intermodulation between the various parts
of the signal, and much of it is probably broadband in nature.

You're never going to be able to process this so that it sounds
anywhere nearly as good as a 16-bit capture would have sounded. All
of the low-level detail in the signal has been lost due to truncation.
The best you're going to be able to do is make it somewhat less
awful-sounding.

I would guess that you ought to be able to use SoundForge, CoolEdit
(if you have a copy), DART, DC-ART, or any of a number of similar
sound tools to clean up the files somewhat. You'll probably need to
do it in several steps.

What I'd probably try, myself, would be the following:

- Convert the file to 16-bit format at the same sampling rate, to give
yourself some processing headroom.

- There may be a "crackle" filter available - if so, experiment with
it and see if it helps.

- Apply a low-pass filter, to get rid of the higher-frequency
harmonic and intermodulation distortion and hiss. You'll
necessarily be sacrificing some of the treble in the signal if you
do this. Play around with the filtering and equalization, to get
the best tradeoff between noise reduction and loss-of-treble.

- Apply a dynamic noise-reduction filter - one which adapts in real
time to the amplitude of the signal. Some of these are low-pass
filters with automatic adjustment of the filter "knee" frequency
and slope. Others are multi-band filter banks, where each bank
will pass its specified frequency range only when there's more than
a certain amount of signal present.


As a quickie hack, you could try converting one of these files to
16-bit format, and then running it through an MP3 encoder at a
relatively low encoding bit rate. This _might_ cause some of the hiss
and noise to be discarded as "not musically or sonically worth
keeping", and the more musical frequency bands to be retained.

--
Dave Platt AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is there a
program that can clean up the file (IE, remove the hiss and crackle and or
improve overall sound quality)?

Unfortunately, it's likely that you did some very difficult-to-repair
damage to the audio quality. By recording at 8 bits of resolution,
and not having a proper "dither" in the process, you will have
generated a great deal of quantization noise in the signal. This
noise (the crackling and hiss that you hear) is, mathematically, quite
a mess... some of it may be harmonically related to the original
signal, some of it will be intermodulation between the various parts
of the signal, and much of it is probably broadband in nature.

You're never going to be able to process this so that it sounds
anywhere nearly as good as a 16-bit capture would have sounded. All
of the low-level detail in the signal has been lost due to truncation.
The best you're going to be able to do is make it somewhat less
awful-sounding.

I would guess that you ought to be able to use SoundForge, CoolEdit
(if you have a copy), DART, DC-ART, or any of a number of similar
sound tools to clean up the files somewhat. You'll probably need to
do it in several steps.

What I'd probably try, myself, would be the following:

- Convert the file to 16-bit format at the same sampling rate, to give
yourself some processing headroom.

- There may be a "crackle" filter available - if so, experiment with
it and see if it helps.

- Apply a low-pass filter, to get rid of the higher-frequency
harmonic and intermodulation distortion and hiss. You'll
necessarily be sacrificing some of the treble in the signal if you
do this. Play around with the filtering and equalization, to get
the best tradeoff between noise reduction and loss-of-treble.

- Apply a dynamic noise-reduction filter - one which adapts in real
time to the amplitude of the signal. Some of these are low-pass
filters with automatic adjustment of the filter "knee" frequency
and slope. Others are multi-band filter banks, where each bank
will pass its specified frequency range only when there's more than
a certain amount of signal present.


As a quickie hack, you could try converting one of these files to
16-bit format, and then running it through an MP3 encoder at a
relatively low encoding bit rate. This _might_ cause some of the hiss
and noise to be discarded as "not musically or sonically worth
keeping", and the more musical frequency bands to be retained.

--
Dave Platt AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

Thank you for such a comprehensive answer. I do have CE pro and will try
what you suggest. Fortunately, This only happened to about 15 out of 800
files, but still needs whatever correction is available. If I could impose
on you for one more (unrelated) issue:

I have an ATI AIW 9000 in one machine from which I occassionally record
cable TV movies (most video I record is done on another machine which has a
Navis Pro card...great card!). The problem with the ATI is that the recorded
sound is very low, even though the volumes used (master, recording and line
in) are all the way up (this seems to be an issue with these cards). Is
there a program that will simply and quickly amplify the volume? It's not
important enough to go through a lengthy process, but would be nice if I
could run the file through a quick process.

Once again, thanks for your time and expertise,

DV



"Dave Platt" wrote in message
...
I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is
there a
program that can clean up the file (IE, remove the hiss and crackle and
or
improve overall sound quality)?

Unfortunately, it's likely that you did some very difficult-to-repair
damage to the audio quality. By recording at 8 bits of resolution,
and not having a proper "dither" in the process, you will have
generated a great deal of quantization noise in the signal. This
noise (the crackling and hiss that you hear) is, mathematically, quite
a mess... some of it may be harmonically related to the original
signal, some of it will be intermodulation between the various parts
of the signal, and much of it is probably broadband in nature.

You're never going to be able to process this so that it sounds
anywhere nearly as good as a 16-bit capture would have sounded. All
of the low-level detail in the signal has been lost due to truncation.
The best you're going to be able to do is make it somewhat less
awful-sounding.

I would guess that you ought to be able to use SoundForge, CoolEdit
(if you have a copy), DART, DC-ART, or any of a number of similar
sound tools to clean up the files somewhat. You'll probably need to
do it in several steps.

What I'd probably try, myself, would be the following:

- Convert the file to 16-bit format at the same sampling rate, to give
yourself some processing headroom.

- There may be a "crackle" filter available - if so, experiment with
it and see if it helps.

- Apply a low-pass filter, to get rid of the higher-frequency
harmonic and intermodulation distortion and hiss. You'll
necessarily be sacrificing some of the treble in the signal if you
do this. Play around with the filtering and equalization, to get
the best tradeoff between noise reduction and loss-of-treble.

- Apply a dynamic noise-reduction filter - one which adapts in real
time to the amplitude of the signal. Some of these are low-pass
filters with automatic adjustment of the filter "knee" frequency
and slope. Others are multi-band filter banks, where each bank
will pass its specified frequency range only when there's more than
a certain amount of signal present.


As a quickie hack, you could try converting one of these files to
16-bit format, and then running it through an MP3 encoder at a
relatively low encoding bit rate. This _might_ cause some of the hiss
and noise to be discarded as "not musically or sonically worth
keeping", and the more musical frequency bands to be retained.

--
Dave Platt AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

Thank you for such a comprehensive answer. I do have CE pro and will try
what you suggest. Fortunately, This only happened to about 15 out of 800
files, but still needs whatever correction is available. If I could impose
on you for one more (unrelated) issue:

I have an ATI AIW 9000 in one machine from which I occassionally record
cable TV movies (most video I record is done on another machine which has a
Navis Pro card...great card!). The problem with the ATI is that the recorded
sound is very low, even though the volumes used (master, recording and line
in) are all the way up (this seems to be an issue with these cards). Is
there a program that will simply and quickly amplify the volume? It's not
important enough to go through a lengthy process, but would be nice if I
could run the file through a quick process.

Once again, thanks for your time and expertise,

DV



"Dave Platt" wrote in message
...
I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is
there a
program that can clean up the file (IE, remove the hiss and crackle and
or
improve overall sound quality)?

Unfortunately, it's likely that you did some very difficult-to-repair
damage to the audio quality. By recording at 8 bits of resolution,
and not having a proper "dither" in the process, you will have
generated a great deal of quantization noise in the signal. This
noise (the crackling and hiss that you hear) is, mathematically, quite
a mess... some of it may be harmonically related to the original
signal, some of it will be intermodulation between the various parts
of the signal, and much of it is probably broadband in nature.

You're never going to be able to process this so that it sounds
anywhere nearly as good as a 16-bit capture would have sounded. All
of the low-level detail in the signal has been lost due to truncation.
The best you're going to be able to do is make it somewhat less
awful-sounding.

I would guess that you ought to be able to use SoundForge, CoolEdit
(if you have a copy), DART, DC-ART, or any of a number of similar
sound tools to clean up the files somewhat. You'll probably need to
do it in several steps.

What I'd probably try, myself, would be the following:

- Convert the file to 16-bit format at the same sampling rate, to give
yourself some processing headroom.

- There may be a "crackle" filter available - if so, experiment with
it and see if it helps.

- Apply a low-pass filter, to get rid of the higher-frequency
harmonic and intermodulation distortion and hiss. You'll
necessarily be sacrificing some of the treble in the signal if you
do this. Play around with the filtering and equalization, to get
the best tradeoff between noise reduction and loss-of-treble.

- Apply a dynamic noise-reduction filter - one which adapts in real
time to the amplitude of the signal. Some of these are low-pass
filters with automatic adjustment of the filter "knee" frequency
and slope. Others are multi-band filter banks, where each bank
will pass its specified frequency range only when there's more than
a certain amount of signal present.


As a quickie hack, you could try converting one of these files to
16-bit format, and then running it through an MP3 encoder at a
relatively low encoding bit rate. This _might_ cause some of the hiss
and noise to be discarded as "not musically or sonically worth
keeping", and the more musical frequency bands to be retained.

--
Dave Platt AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

Thank you for such a comprehensive answer. I do have CE pro and will try
what you suggest. Fortunately, This only happened to about 15 out of 800
files, but still needs whatever correction is available. If I could impose
on you for one more (unrelated) issue:

I have an ATI AIW 9000 in one machine from which I occassionally record
cable TV movies (most video I record is done on another machine which has a
Navis Pro card...great card!). The problem with the ATI is that the recorded
sound is very low, even though the volumes used (master, recording and line
in) are all the way up (this seems to be an issue with these cards). Is
there a program that will simply and quickly amplify the volume? It's not
important enough to go through a lengthy process, but would be nice if I
could run the file through a quick process.

Once again, thanks for your time and expertise,

DV



"Dave Platt" wrote in message
...
I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is
there a
program that can clean up the file (IE, remove the hiss and crackle and
or
improve overall sound quality)?

Unfortunately, it's likely that you did some very difficult-to-repair
damage to the audio quality. By recording at 8 bits of resolution,
and not having a proper "dither" in the process, you will have
generated a great deal of quantization noise in the signal. This
noise (the crackling and hiss that you hear) is, mathematically, quite
a mess... some of it may be harmonically related to the original
signal, some of it will be intermodulation between the various parts
of the signal, and much of it is probably broadband in nature.

You're never going to be able to process this so that it sounds
anywhere nearly as good as a 16-bit capture would have sounded. All
of the low-level detail in the signal has been lost due to truncation.
The best you're going to be able to do is make it somewhat less
awful-sounding.

I would guess that you ought to be able to use SoundForge, CoolEdit
(if you have a copy), DART, DC-ART, or any of a number of similar
sound tools to clean up the files somewhat. You'll probably need to
do it in several steps.

What I'd probably try, myself, would be the following:

- Convert the file to 16-bit format at the same sampling rate, to give
yourself some processing headroom.

- There may be a "crackle" filter available - if so, experiment with
it and see if it helps.

- Apply a low-pass filter, to get rid of the higher-frequency
harmonic and intermodulation distortion and hiss. You'll
necessarily be sacrificing some of the treble in the signal if you
do this. Play around with the filtering and equalization, to get
the best tradeoff between noise reduction and loss-of-treble.

- Apply a dynamic noise-reduction filter - one which adapts in real
time to the amplitude of the signal. Some of these are low-pass
filters with automatic adjustment of the filter "knee" frequency
and slope. Others are multi-band filter banks, where each bank
will pass its specified frequency range only when there's more than
a certain amount of signal present.


As a quickie hack, you could try converting one of these files to
16-bit format, and then running it through an MP3 encoder at a
relatively low encoding bit rate. This _might_ cause some of the hiss
and noise to be discarded as "not musically or sonically worth
keeping", and the more musical frequency bands to be retained.

--
Dave Platt AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

Thank you for such a comprehensive answer. I do have CE pro and will try
what you suggest. Fortunately, This only happened to about 15 out of 800
files, but still needs whatever correction is available. If I could impose
on you for one more (unrelated) issue:

I have an ATI AIW 9000 in one machine from which I occassionally record
cable TV movies (most video I record is done on another machine which has a
Navis Pro card...great card!). The problem with the ATI is that the recorded
sound is very low, even though the volumes used (master, recording and line
in) are all the way up (this seems to be an issue with these cards). Is
there a program that will simply and quickly amplify the volume? It's not
important enough to go through a lengthy process, but would be nice if I
could run the file through a quick process.

Once again, thanks for your time and expertise,

DV



"Dave Platt" wrote in message
...
I have some .wav files that I intended to be recorded at 22,050 16 bit
stereo but were mistakenly batch recorded at 22,050 8 bit stereo. Is
there a
program that can clean up the file (IE, remove the hiss and crackle and
or
improve overall sound quality)?

Unfortunately, it's likely that you did some very difficult-to-repair
damage to the audio quality. By recording at 8 bits of resolution,
and not having a proper "dither" in the process, you will have
generated a great deal of quantization noise in the signal. This
noise (the crackling and hiss that you hear) is, mathematically, quite
a mess... some of it may be harmonically related to the original
signal, some of it will be intermodulation between the various parts
of the signal, and much of it is probably broadband in nature.

You're never going to be able to process this so that it sounds
anywhere nearly as good as a 16-bit capture would have sounded. All
of the low-level detail in the signal has been lost due to truncation.
The best you're going to be able to do is make it somewhat less
awful-sounding.

I would guess that you ought to be able to use SoundForge, CoolEdit
(if you have a copy), DART, DC-ART, or any of a number of similar
sound tools to clean up the files somewhat. You'll probably need to
do it in several steps.

What I'd probably try, myself, would be the following:

- Convert the file to 16-bit format at the same sampling rate, to give
yourself some processing headroom.

- There may be a "crackle" filter available - if so, experiment with
it and see if it helps.

- Apply a low-pass filter, to get rid of the higher-frequency
harmonic and intermodulation distortion and hiss. You'll
necessarily be sacrificing some of the treble in the signal if you
do this. Play around with the filtering and equalization, to get
the best tradeoff between noise reduction and loss-of-treble.

- Apply a dynamic noise-reduction filter - one which adapts in real
time to the amplitude of the signal. Some of these are low-pass
filters with automatic adjustment of the filter "knee" frequency
and slope. Others are multi-band filter banks, where each bank
will pass its specified frequency range only when there's more than
a certain amount of signal present.


As a quickie hack, you could try converting one of these files to
16-bit format, and then running it through an MP3 encoder at a
relatively low encoding bit rate. This _might_ cause some of the hiss
and noise to be discarded as "not musically or sonically worth
keeping", and the more musical frequency bands to be retained.

--
Dave Platt AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

Also, one more question related to .wav file recordings:

What quality difference can be expected from the 22,050mz 16 bit and 16,000
16 bit? 12,000 16 bit?

Again, thanks for your time and patience,

DV

Also, one more question related to .wav file recordings:

What quality difference can be expected from the 22,050mz 16 bit and 16,000
16 bit? 12,000 16 bit?

Again, thanks for your time and patience,

DV

Also, one more question related to .wav file recordings:

What quality difference can be expected from the 22,050mz 16 bit and 16,000
16 bit? 12,000 16 bit?

Again, thanks for your time and patience,

DV

Also, one more question related to .wav file recordings:

What quality difference can be expected from the 22,050mz 16 bit and 16,000
16 bit? 12,000 16 bit?

Again, thanks for your time and patience,

DV

Dracir Venostos wrote:

Also, one more question related to .wav file recordings:

What quality difference can be expected from the 22,050mz 16 bit and 16,000
16 bit? 12,000 16 bit?

Again, thanks for your time and patience,

DV


Bit depth, as in 16 bits, determines dynamic range, the maximum
difference between the loudest and softest passages of the audio sample.
There's a formula which translates bit depth to dynamic range, but
generally it's 6 db per bit. So 16 bits gives you about 16*6=96 db of
dynamic range. That's the theoretical maximum for 16 bits. Count on your
real world experience to be lower, like 90 db. The sampling frequency,
like 22,050 Hz, determines what frequency spectrum you'll be able to
record. At 22,050 Hz, you can record frequencies up to 11,025 Hz, half
the sampling frequency. This half frequnecy is known as ther Nyquist
frequency So, for a 16/12000 recording, your dynamic range is the same
as before, 96 db, but your maximum recordable audio frequency will be
lower, to 6000 Hz.


CD

Dracir Venostos wrote:

Also, one more question related to .wav file recordings:

What quality difference can be expected from the 22,050mz 16 bit and 16,000
16 bit? 12,000 16 bit?

Again, thanks for your time and patience,

DV


Bit depth, as in 16 bits, determines dynamic range, the maximum
difference between the loudest and softest passages of the audio sample.
There's a formula which translates bit depth to dynamic range, but
generally it's 6 db per bit. So 16 bits gives you about 16*6=96 db of
dynamic range. That's the theoretical maximum for 16 bits. Count on your
real world experience to be lower, like 90 db. The sampling frequency,
like 22,050 Hz, determines what frequency spectrum you'll be able to
record. At 22,050 Hz, you can record frequencies up to 11,025 Hz, half
the sampling frequency. This half frequnecy is known as ther Nyquist
frequency So, for a 16/12000 recording, your dynamic range is the same
as before, 96 db, but your maximum recordable audio frequency will be
lower, to 6000 Hz.


CD

Dracir Venostos wrote:

Also, one more question related to .wav file recordings:

What quality difference can be expected from the 22,050mz 16 bit and 16,000
16 bit? 12,000 16 bit?

Again, thanks for your time and patience,

DV


Bit depth, as in 16 bits, determines dynamic range, the maximum
difference between the loudest and softest passages of the audio sample.
There's a formula which translates bit depth to dynamic range, but
generally it's 6 db per bit. So 16 bits gives you about 16*6=96 db of
dynamic range. That's the theoretical maximum for 16 bits. Count on your
real world experience to be lower, like 90 db. The sampling frequency,
like 22,050 Hz, determines what frequency spectrum you'll be able to
record. At 22,050 Hz, you can record frequencies up to 11,025 Hz, half
the sampling frequency. This half frequnecy is known as ther Nyquist
frequency So, for a 16/12000 recording, your dynamic range is the same
as before, 96 db, but your maximum recordable audio frequency will be
lower, to 6000 Hz.


CD

Dracir Venostos wrote:

Also, one more question related to .wav file recordings:

What quality difference can be expected from the 22,050mz 16 bit and 16,000
16 bit? 12,000 16 bit?

Again, thanks for your time and patience,

DV


Bit depth, as in 16 bits, determines dynamic range, the maximum
difference between the loudest and softest passages of the audio sample.
There's a formula which translates bit depth to dynamic range, but
generally it's 6 db per bit. So 16 bits gives you about 16*6=96 db of
dynamic range. That's the theoretical maximum for 16 bits. Count on your
real world experience to be lower, like 90 db. The sampling frequency,
like 22,050 Hz, determines what frequency spectrum you'll be able to
record. At 22,050 Hz, you can record frequencies up to 11,025 Hz, half
the sampling frequency. This half frequnecy is known as ther Nyquist
frequency So, for a 16/12000 recording, your dynamic range is the same
as before, 96 db, but your maximum recordable audio frequency will be
lower, to 6000 Hz.


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