Linear PCM audio: 44.1 KHz sample-rate, monaural, 22,050-bits-per-second

Discussion in 'Computer Support' started by Radium, Jul 23, 2007.

  1. Radium

    Radium Guest

    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". New stuff is learned
    everyday.

    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 incorrect 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
     
    Radium, Jul 23, 2007
    #1
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  2. Radium

    Jordan Guest

    Hi Radium

    You need double the sampling rate to reproduce an analogue waveform,
    according to Nyqvist theorem.
     
    Jordan, Jul 23, 2007
    #2
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  3. Correct. The 20k sampling rate would yield a maximum of 10k waveforms,
    which for older folks and people with hearing damage would pretty much
    cover the audio spectrum.
     
    =?ISO-8859-1?Q?R=F4g=EAr?=, Jul 23, 2007
    #3
  4. Radium

    GHalleck Guest

    True, but only if there was a sharp cutoff at 10 KHz, which is not the
    case. The overtones above 10 KHz, which is normally present in analog,
    would be lost under digital at a 20,000-bit sampling rate.

    Even today, the music industry regrets having set the 44.1 KHz sampling
    rate as the standard. If the musicians and listeners had their way, the
    sampling rate would have been much higher, perhaps 68 KHz, just to match
    the quality of analog releases.
     
    GHalleck, Jul 23, 2007
    #4
  5. Not necessarily. In modern signal processing, signals are decomposed
    into components by an analysis phase, say, by Fourier Transforms (while
    more often in its special case of the Discrete Cosine Transform, for a
    lot of good theoretical and practical reasons) or by other means like
    wavelet transforms. The signal is then separated into its components
    and it is the information about the components that is coded.

    A complex sound wave might be the result of adding two basic frequencies
    with given amplitudes. One would then code the frequency and amplitude
    of these two frequencies to represent the signal. If the description is
    "I have frequency F1 with amplitude A1 combined with F2 with A2, over
    1 second", the number of bits per sample is very, very, low, achieving
    a high compression ratio.

    In the real world, however, sound waves are the results of a great
    number of frequency/amplitude and they are not totally periodic.
    Algorithms will decompose the signal into the component frequencies,
    code each to a certain precision, thus yielding an acceptable
    reconstruction at decompression. The efficiency of the compression
    therefore depends on how you decompose the signal and how smart you are
    about selectively destroying precision so that the signal is compressed
    to a certain amount of bits.


    Best,

    S.
     
    Steven Pigeon, Jul 23, 2007
    #5
  6. Radium

    Phil Carmody Guest

    Nonsense. You can digitise anything, such as the human voice
    using a 1-bit signal wave. I forget the name of the technique
    now, something like delta modulation. I did it back on my ZX
    Spectrum in the early 80s. Basically, if the gradient of the
    signal is negative output a 0, and if it's positive output a
    1. It's crummy, but it's recognisable.

    The only reason you'd get a 'tick-tock' is if you were trying
    to digitise something that vaguely resembled a 'tick-tock' sound.

    Phil
     
    Phil Carmody, Jul 23, 2007
    #6
  7. Me too.
     
    =?ISO-8859-1?Q?R=F4g=EAr?=, Jul 23, 2007
    #7
  8. Radium

    Don Pearce Guest

    Sort of right. What happens is this. The 1/0 digital signal is used to
    charge or discharge a capacitor. At each time tick, the voltage on
    that capacitor is compared to the level of the audio signal. If it is
    too low, a 1 will charge it up a bit, and if it is too high a 0 will
    drain a little charge from it, keeping it matched exactly to the
    audio. The resulting stream of 1s and 0s describes the audio signal.

    d
     
    Don Pearce, Jul 23, 2007
    #8
  9. Radium

    Pete Fraser Guest

    It's only crummy if you don't sample fast enough.
    Many audio A/D converters work like this.
     
    Pete Fraser, Jul 23, 2007
    #9
  10. Radium

    Karl Uppiano Guest

    Practically all of the current Sigma/Delta (or Delta/Sigma) converters do
    this. They are sometimes called MASH converters or 1-bit converters. If the
    1-bit converter sample fast enough (in the MHz range) that resolution is not
    lost at the maximum slew-rate (which can be calculated from the slope of the
    desired maximum input frequency at full scale amplitude), then it is
    theoretically capable of arbitrarily high resolution (i.e., quality). The
    output of this converter is then numerically transformed into a 16-bit or
    24-bit PCM datastream at the selected sample rate (e.g., 44.1KHz or
    whatever).

    A well-made Sigma/Delta converter can be made more reliable than the older
    successive approximation register (SAR) converters, because of the
    difficulty of trimming SAR resistors or capacitors to the exacting precision
    needed for 16 or 24 bit audio (all 24 voltage references must be accurate to
    about 3 millionths of a percent for 24 bit accuracy to 1/2 LSB - impossible,
    even if hand trimmed - and it wouldn't stay that way for long!). A 1-bit
    converter, on the other hand, only requires one coarsely-trimmed voltage
    reference, and accurate timing, which is much easier to achieve with careful
    circuit layout.

    http://www.maxim-ic.com/appnotes.cfm/appnote_number/1870
     
    Karl Uppiano, Jul 24, 2007
    #10
  11. Radium

    Jasen Betts Guest

    I recall hearing speech from the games "Ghost Busters" and "Freedom
    Fighter"

    In the 90s modplay would overlay wav files an play them out the PC
    speaker (which is also one-bit).

    Bye.
    Jasen
     
    Jasen Betts, Jul 24, 2007
    #11
  12. Radium

    Willem Guest

    Jasen wrote:
    )> Nonsense. You can digitise anything, such as the human voice
    )> using a 1-bit signal wave. I forget the name of the technique
    )> now,
    )
    ) delta-sigma.
    )
    )> something like delta modulation. I did it back on my ZX
    )> Spectrum in the early 80s. Basically, if the gradient of the
    )> signal is negative output a 0, and if it's positive output a
    )> 1. It's crummy, but it's recognisable.
    )
    ) I recall hearing speech from the games "Ghost Busters" and "Freedom
    ) Fighter"
    )
    ) In the 90s modplay would overlay wav files an play them out the PC
    ) speaker (which is also one-bit).

    But that's a different technique. You can program the PC speaker to putput
    a square wave with a programmed phase width and a high frequency, and then
    vary the phase width you get different samples.


    SaSW, Willem
    --
    Disclaimer: I am in no way responsible for any of the statements
    made in the above text. For all I know I might be
    drugged or something..
    No I'm not paranoid. You all think I'm paranoid, don't you !
    #EOT
     
    Willem, Jul 24, 2007
    #12
  13. LOL. Willem, I wanna 'putput' you into a jar, display it to the masses
    and say "This, folks, is why we can't have nice things!"
     
    industrial_one, Jul 24, 2007
    #13
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