Linear PCM audio: 44.1 KHz, monaural, 1-bit-per-second

Radium <glucegen1@gmail.com> writes:

> On Jul 22, 12:37 pm, glen herrmannsfeldt <g...@ugcs.caltech.edu>
> wrote:
>
>> I read it as an average rate of 1 bit/second, after some compression
>> method. I make no suggestion on what compression method might
>> be able to do that.
>
> Any help here? Is there any type of compression that most closely
> resembles linear-PCM and can do the task of less than 1-bit-per-cycle?
>
> One-bit-per-44,100-cycles is what I was originally looking for.
> However, as some posters have stated, all I would hear in 1-bit-per-
> cycle would resemble a square wave "tick tock".
>
> So how about decreasing the amount of bits-per-cycle so that the bit-
> rate becomes 20,000-bits-per-second? After all the human auditory
> system perceives up to 20 KHz so covering the entire human audio
> frequency range would require at least 20-kilobits-per-second.
>
> In a sample rate of 44,100-cycles-per-second, this would best be done
> at 1-bit-every-2-cycles. This would give a bit-rate of 22,050-bits-per-
> second. That's obviously above 20kbits-per-second but only slightly.

Radium,

Here is what I suggest to you:

1. Stop wasting your time and get back to your high school
studies.

2. When you finish high school, find a good engineering
college and enroll in the electrical engineering curriculum.

3. When you've finished your EE undergraduate degree, continue
your studies in a masters or PhD curriculum specializing in
DSP and communications.

4. When you've completed your graduate DSP classes and perhaps had
an advanced class in compression or information theory, come back
and discuss your questions with us.

Until you've reached this point, you're unprepared to seriously
discuss this subject.
--
% Randy Yates % "She has an IQ of 1001, she has a jumpsuit
%% Fuquay-Varina, NC % on, and she's also a telephone."
%%% 919-577-9882 %
%%%% <yates@ieee.org> % 'Yours Truly, 2095', *Time*, ELO
http://home.earthlink.net/~yatescr

On Sun, 22 Jul 2007 12:55:24 -0400, Jerry Avins wrote:

> I wasn't addressing that part of the thread. He wants to represent
> 44,100 *uncompressed* sample with a single bit. He seems to feel that
> each sample can be represented by 1/44,100th of a bit.

Well it can be, if the samples aren't very interesting.

--
Andrew

On Sun, 22 Jul 2007 16:18:00 -0400, Jerry Avins wrote:

> I repeat, uncompressed and only one bit for every 44,100 samples.

The misconception, on his part, is not that what he describes is
impossible (it is), but that it can represent audio. Audio (for humans)
has certain bandwidth and (I suspect) information rate boundaries.

--
Andrew

On Sun, 22 Jul 2007 11:22:01 -0400, Jerry Avins wrote:

> On the whole, by a particular accounting. How could you represent part
> of a single bit and no part of any other?

Why would you want to?

The original poster mentioned a bit rate, over an indeterminate period.
I.e., an indeterminate number of bits in total. There was no obvious
requirement, for instance, that the encoding be able to represent fewer
than 44100 samples.

--
Andrew

On Sun, 22 Jul 2007 22:00:48 +0200, andre wrote:

> Radium wrote:
>> what would such audio sound like? Bad-quality?
>
> tic tac tic tac .....

Or some variation on that. Doesn't have to be a boring as a square wave.
Could be encoding centre frequency of a two-tone oscilator (ambulance
simulator with syncopation), or encoding a walking tone,
differential-style. Your one bit/second could be driving a random walk
(i.e. ADPCM of some sort) that controls the characteristics of a filter
with a white noise input. Still not going to be particularly interesting,
and the out-of-band data required to describe the process would no doubt
dwarf the encoding :-)

Cheers,

--
Andrew

On Jul 20, 6:08 pm, Oli Charlesworth <ca...@olifilth.co.uk> wrote:

> This makes no sense. How can you describe 44100 samples with a single
> bit?

Well, one-bit-per-44,100-cycles is what I was originally looking for.
However, as some posters have stated, all I would hear in 1-bit-per-
cycle would resemble a square wave "tick tock".

So how about decreasing the amount of bits-per-cycle so that the bit-
rate becomes 20,000-bits-per-second? After all the human auditory
system perceives up to 20 KHz so covering the entire human audio
frequency range would require at least 20-kilobits-per-second.

In a sample rate of 44,100-cycles-per-second, this would best be done
at 1-bit-every-2-cycles. This would give a bit-rate of 22,050-bits-per-
second. That's obviously above 20kbits-per-second but only slightly.

Couldn't the bit-rate be less than the sample-rate if some information
in each sample is thrown away? Could this data-reduction be done
linearly?

With each 44,100-cycles-per-second, discard half the information, and
you get 22,050-bits-per-second. Discard 1 bit for every two cycles. I
could be wrong though. If so, please assist me.

To all:

I have a neurological disability called Asperger's Syndrome.

I would like to give you some information about my disability. The
reason I am posting this message about Asperger's is to help avoid any
potential misunderstandings [though it's probably too late].

I have been diagnosed with Asperger's Syndrome (AS). AS is a
neurological condition that causes significant impairment in social
interactions. People with AS see the world differently and this can
often bring them in conflict with conventional ways of thinking. They
have difficulty in reading body language, and interpreting subtle
cues. In my situation, I have significant difficulty with natural
conversation, reading social cues, and maintaining eye contact. This
can lead to a great deal of misunderstanding about my intent or my
behavior. For example, I may not always know what to say in social
situations, so I may look away or may not say anything. I also may not
always respond quickly when asked direct questions, but if given time
I am able express my ideas.

On Usenet, the text-equivalent of my disability is probably noticed. I
do apologize profusely, for any inconvenience it causes.

Thank you very much in advance for your understanding, cooperation,
and assistance.


Regards,

Radium

Andrew Reilly wrote:
> On Sun, 22 Jul 2007 11:22:01 -0400, Jerry Avins wrote:
>
>> On the whole, by a particular accounting. How could you represent part
>> of a single bit and no part of any other?
>
> Why would you want to?
>
> The original poster mentioned a bit rate, over an indeterminate period.
> I.e., an indeterminate number of bits in total. There was no obvious
> requirement, for instance, that the encoding be able to represent fewer
> than 44100 samples.

How do you parse "The 1-bit-per-second is because the bit-resolution
[normally 16-bit in CD audio] is only 1-bit-per-44,100 samples in this
hypothetical case"? How many bits per sample does that come to? How is
each sample represented? I'm through with this silly thread.

Jerry
--
Engineering is the art of making what you want from things you can get.
¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯

On Sun, 22 Jul 2007 22:30:12 -0400, Jerry Avins wrote:

> Andrew Reilly wrote:
>> On Sun, 22 Jul 2007 11:22:01 -0400, Jerry Avins wrote:
>>
>>> On the whole, by a particular accounting. How could you represent part
>>> of a single bit and no part of any other?
>>
>> Why would you want to?
>>
>> The original poster mentioned a bit rate, over an indeterminate period.
>> I.e., an indeterminate number of bits in total. There was no obvious
>> requirement, for instance, that the encoding be able to represent fewer
>> than 44100 samples.
>
> How do you parse "The 1-bit-per-second is because the bit-resolution
> [normally 16-bit in CD audio] is only 1-bit-per-44,100 samples in this
> hypothetical case"?

I parse that as "44100 samples, in one second, are encoded with one bit
[and I know that it is usually 16-bit per sample in CD audio]"

> How many bits per sample does that come to?

He is not proposing to encode individual samples. Only chunks of 44100 of
them.

> How is
> each sample represented? I'm through with this silly thread.

That's fine. It certainly is starting from a silly premise, but I've only
joined in to help those who want to emphasise that there is not
necessarily any relationship between the sample rate or the quantization
level of an audio system and the bit-rate of the encoded form. The latter
only has to cover perceptable information rates (i.e., after filtering for
redundancy and relevance). As I said in another post, an information rate
of one bit/second is unlikely to be able to encode anythign that a human
would find very interesting, and probably wouldn't call "audio", but there
are certainly some audio encodings that start to get interesting around 1k
bits/second (voice specific). By about 8k bits/second you can certainly
call it "audio" (that's about what GSM mobile phone gives you). By around
30k to 60k bits/second you can get lo-fi stereo that is certainly "audio".
Pretty much all of the bits/second beyond there are devoted to either
simpler algorithms that don't find as much redundancy in the source, or
different trade-offs in what is considered "relevant" for human listeners.

Note that for all of the bit-rates that I've just mentioned, covering
1kb/s to around 1.5Mb/s (maybe even higher, these days), the result can
be represented as 48kHz-sampled audio. There isn't a relationship between
bits/s and samples/s.

Cheers,

--
Andrew

Andrew Reilly <andrew-newspost@areilly.bpc-users.org> writes:
> [...]
> I've only joined in to help those who want to emphasise that there
> is not necessarily any relationship between the sample rate or the
> quantization level of an audio system and the bit-rate of the
> encoded form.

Show me a *linear PCM* system in which the bit-rate is not directly related
to the sample rate and sample width.
--
% Randy Yates % "She's sweet on Wagner-I think she'd die for Beethoven.
%% Fuquay-Varina, NC % She love the way Puccini lays down a tune, and
%%% 919-577-9882 % Verdi's always creepin' from her room."
%%%% <yates@ieee.org> % "Rockaria", *A New World Record*, ELO
http://home.earthlink.net/~yatescr

On Sun, 22 Jul 2007 23:52:26 -0400, Randy Yates wrote:

> Andrew Reilly <andrew-newspost@areilly.bpc-users.org> writes:
>> [...]
>> I've only joined in to help those who want to emphasise that there
>> is not necessarily any relationship between the sample rate or the
>> quantization level of an audio system and the bit-rate of the
>> encoded form.
>
> Show me a *linear PCM* system in which the bit-rate is not directly related
> to the sample rate and sample width.

How is that relevant? By weaseling in with "sample width", you've already
admitted the possiblity of scaling the bit rate independent of the sample
rate. DSD is linear PCM, after all...

Cheers,

--
Andrew