1-Bit Wave File?

Archived from groups: rec.audio.tech,comp.dsp (More info?)

I am looking for audio software that allows conversion of 16-bit WAVs
and 8-bit WAVs to 1-bit WAVs. I have used Adobe Audition and CakeWalk
Pyro. Neither of them work. CakeWalk does have a "bit-depth converter"
as a FX, however, when I try to use it, I get a runtime error and
Cakewalk automatically closes.
54 answers Last reply
More about wave file
  1. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    curious11112001@yahoo.com (Curious) writes:

    > I am looking for audio software that allows conversion of 16-bit WAVs
    > and 8-bit WAVs to 1-bit WAVs.

    Why do you want to convert to a 1-bit WAV?
    --
    % Randy Yates % "Maybe one day I'll feel her cold embrace,
    %% Fuquay-Varina, NC % and kiss her interface,
    %%% 919-577-9882 % til then, I'll leave her alone."
    %%%% <yates@ieee.org> % 'Yours Truly, 2095', *Time*, ELO
    http://home.earthlink.net/~yatescr
  2. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    On 16 May 2004 14:30:43 -0700, curious11112001@yahoo.com (Curious)
    wrote:

    >I am looking for audio software that allows conversion of 16-bit WAVs
    >and 8-bit WAVs to 1-bit WAVs. I have used Adobe Audition and CakeWalk
    >Pyro. Neither of them work. CakeWalk does have a "bit-depth converter"
    >as a FX, however, when I try to use it, I get a runtime error and
    >Cakewalk automatically closes.


    The wave editor in Magix's mp3 maker can do that. But all you get is
    noise. Do you understand what bit depth is all about?

    Abbedd
  3. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Curious wrote:

    > I am looking for audio software that allows conversion of 16-bit WAVs
    > and 8-bit WAVs to 1-bit WAVs. I have used Adobe Audition and CakeWalk
    > Pyro. Neither of them work. CakeWalk does have a "bit-depth converter"
    > as a FX, however, when I try to use it, I get a runtime error and
    > Cakewalk automatically closes.

    Do you really want a 1-bit on/off signal or did you mean to ask about
    1-bit delta sigma modulation?
  4. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    On 2004-05-16, Curious <curious11112001@yahoo.com> wrote:
    > I am looking for audio software that allows conversion of 16-bit WAVs
    > and 8-bit WAVs to 1-bit WAVs. I have used Adobe Audition and CakeWalk
    > Pyro. Neither of them work. CakeWalk does have a "bit-depth converter"
    > as a FX, however, when I try to use it, I get a runtime error and
    > Cakewalk automatically closes.

    The software in the Roland SP-808 lets you do this...in real time,
    even. You use a knob to select the number of bits. Plus, it lets you
    adjust the sample rate and even whether
    or not you want the noise-shaping filter turned on!

    p.s. I found this link on Randy Yates' page about noise-shaping...

    http://home.earthlink.net/~yatescr/noiseb.ps

    I think a line from that paper may answer Randy's question about
    WHY you want to do this:

    "...quantisation of highly correlated signals (such as music) results
    in tonal distortion components being added to the signal."

    rock and roll,
    -N

    --
    different MP3 every day! http://gweep.net/~shifty/snackmaster
    . . . . . . . . ... . . . . . .
    "Maybe if you ever picked up a goddamn keyboard | Niente
    and compiler, you'd know yourself." -Matthew 7:1 | shifty@gweep.net
  5. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Tachyon <shiftyATATATsidehack.sat.gweep.net> writes:
    > [...]
    > p.s. I found this link on Randy Yates' page about noise-shaping...
    >
    > http://home.earthlink.net/~yatescr/noiseb.ps
    >
    > I think a line from that paper may answer Randy's question about
    > WHY you want to do this:
    >
    > "...quantisation of highly correlated signals (such as music) results
    > in tonal distortion components being added to the signal."

    Hey Tachyon,

    So you're assuming that the OP wants to do this in order to implement
    some sort of musical effect? An interesting possibility.

    My question was intended to be taken at face value. I was asking about
    the OP's intention, not implying that one shouldn't do such a thing.
    My gut feeling is that he wanted to create a 1-bit delta sigma
    bitstream (which won't happen if you merely convert to a 1-bit
    stream), but I don't really know. That's why I asked.

    > rock and roll,
    > -N

    Hey Tachyon, nice screen name. I bet the Romulans don't like you
    too much, though...
    --
    % Randy Yates % "The dreamer, the unwoken fool -
    %% Fuquay-Varina, NC % in dreams, no pain will kiss the brow..."
    %%% 919-577-9882 %
    %%%% <yates@ieee.org> % 'Eldorado Overture', *Eldorado*, ELO
    http://home.earthlink.net/~yatescr
  6. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Tachyon wrote:

    >
    > I think a line from that paper may answer Randy's question about
    > WHY you want to do this:
    >
    > "...quantisation of highly correlated signals (such as music) results
    > in tonal distortion components being added to the signal."


    A little understanding can be a dangerous thing.

    With a 1 bit PCM data stream, there is effectively no recogisable signal. I
    think Randy may have been talking about bit-depths down to around 8 bits.
    However even this quanisation noise is not a type of distortion that is
    likely to sound in any way pleasant in pretty much any style of music.


    geoff
  7. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Geoff Wood" <geoff@paf.co.nz-nospam> wrote in message
    news:eRtqc.2892$FN.302696@news02.tsnz.net...
    > Tachyon wrote:
    >
    > > I think a line from that paper may answer Randy's question about
    > > WHY you want to do this:
    > >
    > > "...quantisation of highly correlated signals (such as music) results
    > > in tonal distortion components being added to the signal."
    >
    > A little understanding can be a dangerous thing.
    >
    > With a 1 bit PCM data stream, there is effectively no recogisable signal. I
    > think Randy may have been talking about bit-depths down to around 8 bits.
    > However even this quanisation noise is not a type of distortion that is
    > likely to sound in any way pleasant in pretty much any style of music.

    Even with 1 bit, if properly dithered, you should be able to recognize the
    signal, though it will be buried in noise. I'm wondering the OP wants to
    convert to some type of SACD (1-bit but very high SR) format?
  8. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    On 2004-05-18, Geoff Wood <geoff@paf.co.nz-nospam> wrote:
    > Tachyon wrote:
    >
    >>
    >> I think a line from that paper may answer Randy's question about
    >> WHY you want to do this:
    >>
    >> "...quantisation of highly correlated signals (such as music) results
    >> in tonal distortion components being added to the signal."
    >
    >
    > A little understanding can be a dangerous thing.
    >
    > With a 1 bit PCM data stream, there is effectively no recogisable signal. I
    > think Randy may have been talking about bit-depths down to around 8 bits.
    > However even this quanisation noise is not a type of distortion that is
    > likely to sound in any way pleasant in pretty much any style of music.

    Fans of the Atari 2600 sound chip TIA beg to differ :) Each of its
    voices is capable of about 6 unique waveforms,
    each of which is a train of 0's and 1's.

    Additionally, square and PWM waveforms, staples of much electronic music,
    are 1-bit waveforms.

    Curiously enough, the TIA can't quite make 50% duty cycle square
    waves, but several close matches, like 15/31 :)

    --
    different MP3 every day! http://gweep.net/~shifty/snackmaster
    . . . . . . . . ... . . . . . .
    "Maybe if you ever picked up a goddamn keyboard | Niente
    and compiler, you'd know yourself." -Matthew 7:1 | shifty@gweep.net
  9. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Tachyon wrote:
    > On 2004-05-18, Geoff Wood <geoff@paf.co.nz-nospam> wrote:
    >
    >>Tachyon wrote:
    >>
    >>
    >>>I think a line from that paper may answer Randy's question about
    >>>WHY you want to do this:
    >>>
    >>>"...quantisation of highly correlated signals (such as music) results
    >>> in tonal distortion components being added to the signal."
    >>
    >>
    >>A little understanding can be a dangerous thing.
    >>
    >>With a 1 bit PCM data stream, there is effectively no recogisable signal. I
    >>think Randy may have been talking about bit-depths down to around 8 bits.
    >>However even this quanisation noise is not a type of distortion that is
    >>likely to sound in any way pleasant in pretty much any style of music.
    >
    >
    > Fans of the Atari 2600 sound chip TIA beg to differ :) Each of its
    > voices is capable of about 6 unique waveforms,
    > each of which is a train of 0's and 1's.
    >
    > Additionally, square and PWM waveforms, staples of much electronic music,
    > are 1-bit waveforms.
    >
    > Curiously enough, the TIA can't quite make 50% duty cycle square
    > waves, but several close matches, like 15/31 :)

    That's all gobbledygook for all the OP cares. He wants to CONVERT .wav
    files to 1-bit form. Either he knows why he wants it, or he doesn't know
    what he wants. Either way, talk of TIA and PWM won't help him.

    Jerry
    --
    Engineering is the art of making what you want from things you can get.
    ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
  10. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    On 2004-05-19, Jerry Avins <jya@ieee.org> wrote:
    > Tachyon wrote:
    >> On 2004-05-18, Geoff Wood <geoff@paf.co.nz-nospam> wrote:
    >>
    >>>Tachyon wrote:
    >>>
    >>>
    >>>>I think a line from that paper may answer Randy's question about
    >>>>WHY you want to do this:
    >>>>
    >>>>"...quantisation of highly correlated signals (such as music) results
    >>>> in tonal distortion components being added to the signal."
    >>>
    >>>
    >>>A little understanding can be a dangerous thing.
    >>>
    >>>With a 1 bit PCM data stream, there is effectively no recogisable signal. I
    >>>think Randy may have been talking about bit-depths down to around 8 bits.
    >>>However even this quanisation noise is not a type of distortion that is
    >>>likely to sound in any way pleasant in pretty much any style of music.
    >>
    >>
    >> Fans of the Atari 2600 sound chip TIA beg to differ :) Each of its
    >> voices is capable of about 6 unique waveforms,
    >> each of which is a train of 0's and 1's.
    >>
    >> Additionally, square and PWM waveforms, staples of much electronic music,
    >> are 1-bit waveforms.
    >>
    >> Curiously enough, the TIA can't quite make 50% duty cycle square
    >> waves, but several close matches, like 15/31 :)
    >
    > That's all gobbledygook for all the OP cares. He wants to CONVERT .wav
    > files to 1-bit form. Either he knows why he wants it, or he doesn't know
    > what he wants. Either way, talk of TIA and PWM won't help him.

    Hey man,

    I'm just sticking up for 1-bit waveforms!


    --
    different MP3 every day! http://gweep.net/~shifty/snackmaster
    . . . . . . . . ... . . . . . .
    "Maybe if you ever picked up a goddamn keyboard | Niente
    and compiler, you'd know yourself." -Matthew 7:1 | shifty@gweep.net
  11. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Randy Yates <yates@ieee.org> wrote in message news:<u0yfes12.fsf@ieee.org>...
    > curious11112001@yahoo.com (Curious) writes:
    >
    > > I am looking for audio software that allows conversion of 16-bit WAVs
    > > and 8-bit WAVs to 1-bit WAVs.
    >
    > Why do you want to convert to a 1-bit WAV?

    I want film-quality sound. By "film quality" I am referring to the
    crackling [resembles the sound of burning coal] and lack of clarity in
    the audio recordings of old B&W films which used the
    variable-intensity recording. I would like to preserve the frequency
    response of CD-audio [keep the 44.1 KHz sampling rate], however, I
    would like to make it mono and film-quality in terms of the artifacts
    mentioned above. For some reason this type of music gets my juices
    going.
  12. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Curious wrote:

    > I am looking for audio software that allows conversion of 16-bit WAVs
    > and 8-bit WAVs to 1-bit WAVs. I have used Adobe Audition and CakeWalk
    > Pyro. Neither of them work. CakeWalk does have a "bit-depth converter"
    > as a FX, however, when I try to use it, I get a runtime error and
    > Cakewalk automatically closes.

    A + Comparator FlipFlop
    I/P -------|\ +------+ Q
    | >----------| D Q|---+---- Bitstream out
    B +--|/ C +--|> | |
    | - | +------+ |
    | | |
    +-/\/\/\-----)-------------+
    | R |
    C === |
    | Fs---+
    0V

    Several years ago I made a delay line to delay the audio for a walkie-talkie
    so that the Tx VOX voice operated switch and Rx unmute operation would be
    completed before the leading word spoken by the operator was transmitted.
    I tried a bucket brigade device (MN3001) but had troubles with noise and
    limitations on maximum delay.

    I tried the cct above as a 1-bit encoder (mentioned in a mag somewhere),
    clocked at over 200KHz. I do not know if the cct has a name.

    The Q o/p will be either high or low during a clk period, such as to force
    the RC LPF voltage at B to trend towards a match with i/p sig A. O/p Q
    tends to be strings of 1's or 0's until match is achieved, then become
    alternating 1's & 0's (50% PWM) until the i/p signal changes again.


    +------+
    BitStream --->-----|D Q|--------> ZQ Delayed BitStream
    | |
    +------+ | |
    Fs----| |---->|/RAS |
    | |---->|/CAS |
    | |---->|/WE |
    | | | |
    | |---->|A0 |
    | PAL |-- |A1 |
    | |-- | " |
    | |-- | " |
    | |-- | " |
    | |---->|An |
    +------+ +------+

    The above cct used a PAL to read the data bit at the current single bit wide
    DRAM chip address, the bit thus read having been written on the previous
    "lap" of the address counter. The PAL then operated /WE to write the
    current Bitstream i/p (0/1) into the current RAM address.

    The number of address lines & clock frequency determine the delay and
    fidelity which can be achieved. In your application this delay line block
    diagram could be implemented by 2 weeks of storage on your hard drive.


    R
    Delayed BitStream ------/\/\/\----+-----> Recovered audio
    |
    === C
    |
    0V

    Recovery of the signal back to analog was by means of another RC LPF of
    identical time constant to the one used in the encoder. The idea is that
    if BitStream Q LPF's to imitate i/p signal A, then the identical but
    delayed BitStream ZQ should LPF to create a signal that looks like A also.


    I did once implement the above with a 56K DSP because I could not get
    samples of the intended ADC chip. It used software (RC LPF's etc.) for
    everything except the comparator (op-amp). It sort of worked, nearly, just
    about, almost, but the high interrupt rate interfered with other
    interrupt-driven processes, causing jitter & noise on the recovered audio.
    It did work well enough to get on with the job until the proper CODEC
    arrived.

    I think that using the above building blocks, you could write a prog to
    encode a WAV file & create a bitstream file. Converting back again should
    be quite simple.

    Jim Adamthwaite.
  13. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Tachyon wrote:
    > On 2004-05-18, Geoff Wood <geoff@paf.co.nz-nospam> wrote:

    > Additionally, square and PWM waveforms, staples of much electronic
    > music, are 1-bit waveforms.

    Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.

    geoff
  14. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Curious" <curious11112001@yahoo.com> wrote in message
    news:34a4f456.0405191922.1d69cdf6@posting.google.com...
    > Randy Yates <yates@ieee.org> wrote in message
    news:<u0yfes12.fsf@ieee.org>...
    > > curious11112001@yahoo.com (Curious) writes:
    > >
    > > > I am looking for audio software that allows conversion of 16-bit WAVs
    > > > and 8-bit WAVs to 1-bit WAVs.
    > >
    > > Why do you want to convert to a 1-bit WAV?
    >
    > I want film-quality sound. By "film quality" I am referring to the
    > crackling [resembles the sound of burning coal]

    Random noise (of relatively specific types).
    Don't see how reducing quantization can accomplish this.
    There are actually recordings of "clean" noise that you
    can mix into your material to simulate this artifact.

    > and lack of clarity in
    > the audio recordings of old B&W films which used the
    > variable-intensity recording.

    Distortion, bandwidth restriction, etc. Distortion might be
    simulated by reducing bit-depth, but likely better ways.

    Of course true "1-bit" audio is indistinguishable from
    random noise.
  15. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Curious wrote:

    > Randy Yates <yates@ieee.org> wrote in message news:<u0yfes12.fsf@ieee.org>...
    >
    >>curious11112001@yahoo.com (Curious) writes:
    >>
    >>
    >>>I am looking for audio software that allows conversion of 16-bit WAVs
    >>>and 8-bit WAVs to 1-bit WAVs.
    >>
    >>Why do you want to convert to a 1-bit WAV?
    >
    >
    > I want film-quality sound. By "film quality" I am referring to the
    > crackling [resembles the sound of burning coal] and lack of clarity in
    > the audio recordings of old B&W films which used the
    > variable-intensity recording. I would like to preserve the frequency
    > response of CD-audio [keep the 44.1 KHz sampling rate], however, I
    > would like to make it mono and film-quality in terms of the artifacts
    > mentioned above. For some reason this type of music gets my juices
    > going.

    You could probably reproduce that effect in more or less the manner it
    was first created by modulating a light bulb and detecting it with a
    photocell. I don't think the film added much but static from dirt,
    and you could probably get that by spraying dust through the beam. It
    actually might be a fun experiment.

    --
    The e-mail address in our reply-to line is reversed in an attempt to
    minimize spam. Our true address is of the form che...@prodigy.net.
  16. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Geoff Wood" <geoff@paf.co.nz-nospam> writes:

    > Geoff Wood wrote:
    >> Tachyon wrote:
    >>> On 2004-05-18, Geoff Wood <geoff@paf.co.nz-nospam> wrote:
    >>
    >>> Additionally, square and PWM waveforms, staples of much electronic
    >>> music, are 1-bit waveforms.
    >>
    >> Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.
    >>
    >
    > On the other hand, I guess it does,

    Absolutely it does not. A PWM signal has the property that at
    time n*T the signal begins at a high state for p*T seconds and
    ends at a low state for (1-p)*T seconds, 0 <= p <= 1. A 1-bit
    bitstream does not have this property.

    Many people seem to have this misconception that a delta sigma
    bitstream is a PWM waveform. It is not - they are two different
    animals.
    --
    % Randy Yates % "I met someone who looks alot like you,
    %% Fuquay-Varina, NC % she does the things you do,
    %%% 919-577-9882 % but she is an IBM."
    %%%% <yates@ieee.org> % 'Yours Truly, 2095', *Time*, ELO
    http://home.earthlink.net/~yatescr
  17. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    curious11112001@yahoo.com (Curious) writes:

    > Randy Yates <yates@ieee.org> wrote in message news:<u0yfes12.fsf@ieee.org>...
    >> curious11112001@yahoo.com (Curious) writes:
    >>
    >> > I am looking for audio software that allows conversion of 16-bit WAVs
    >> > and 8-bit WAVs to 1-bit WAVs.
    >>
    >> Why do you want to convert to a 1-bit WAV?
    >
    > I want film-quality sound. By "film quality" I am referring to the
    > crackling [resembles the sound of burning coal] and lack of clarity in
    > the audio recordings of old B&W films which used the
    > variable-intensity recording. I would like to preserve the frequency
    > response of CD-audio [keep the 44.1 KHz sampling rate], however, I
    > would like to make it mono and film-quality in terms of the artifacts
    > mentioned above. For some reason this type of music gets my juices
    > going.

    Hi Curious,

    That is a realistic goal, and I believe you can achieve it with some
    type of digital signal processing, but I don't believe simply reducing
    the bit-depth will accomplish the effect you're looking for.

    The problem is that the type of noise signal you want has a very
    different character than quantization noise. Quantization noise has a
    uniform probability distribution. What you want will have very large
    "tails" in its distribution. I am speaking here in terms that may be
    unfamiliar to you, but that should be familiar to a DSP engineer.

    I can't recall ever seeing any study or research that characterized
    the noise from a film soundtrack. Robert, have you ever seen this in
    your dealings with the AES?
    --
    % Randy Yates % "Bird, on the wing,
    %% Fuquay-Varina, NC % goes floating by
    %%% 919-577-9882 % but there's a teardrop in his eye..."
    %%%% <yates@ieee.org> % 'One Summer Dream', *Face The Music*, ELO
    http://home.earthlink.net/~yatescr
  18. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Richard Crowley" <rcrowley7@xprt.net> wrote in message
    news:10aola6pnbau090@corp.supernews.com...
    >
    > Of course true "1-bit" audio is indistinguishable from
    > random noise.

    Not really. DSD is one bit (at a very high sampling rate) and certainly doesn't
    sound like random noise. Even with "regular" sample rates such as 44.1kHz, so
    you can still recognize properly-dithered audio at very low bit depths (even 1
    bit). The audio is buried in noise, just like it would sound if you recorded at
    very low levels on a medium such as cassette tape, but the ear is pretty good at
    pulling out the real sound from the noise.
  19. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Curious wrote:

    > Randy Yates <yates@ieee.org> wrote in message news:<u0yfes12.fsf@ieee.org>...
    >
    >>curious11112001@yahoo.com (Curious) writes:
    >>
    >>
    >>>I am looking for audio software that allows conversion of 16-bit WAVs
    >>>and 8-bit WAVs to 1-bit WAVs.
    >>
    >>Why do you want to convert to a 1-bit WAV?
    >
    >
    > I want film-quality sound. By "film quality" I am referring to the
    > crackling [resembles the sound of burning coal] and lack of clarity in
    > the audio recordings of old B&W films which used the
    > variable-intensity recording. I would like to preserve the frequency
    > response of CD-audio [keep the 44.1 KHz sampling rate], however, I
    > would like to make it mono and film-quality in terms of the artifacts
    > mentioned above. For some reason this type of music gets my juices
    > going.

    You may be able to get this effect with a program like CoolEditPro which
    can merge a wide variety of distortion effects into your original sound.
    One of the built-in distortion settings may produce the type of sound
    you are after or you can create your own distortion.

    If you own one of these noisy films, record the sound from a section of
    film that has mostly noise and no audio. A program like CoolEditPro can
    extract the noise from the film section and merge it into any other
    sound file you have.
  20. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Geoff Wood wrote:
    > Tachyon wrote:
    >> On 2004-05-18, Geoff Wood <geoff@paf.co.nz-nospam> wrote:
    >
    >> Additionally, square and PWM waveforms, staples of much electronic
    >> music, are 1-bit waveforms.
    >
    > Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.
    >

    On the other hand, I guess it does, but not a PWM signal that has any
    relationship to the PCM-encoded waveform !

    ;-)

    geoff
  21. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Richard Crowley" <rcrowley7@xprt.net> wrote in message news:<10aola6pnbau090@corp.supernews.com>...
    >
    > Of course true "1-bit" audio is indistinguishable from
    > random noise.
    >

    In a word, wrong. In two words, completely wrong.

    A 1-bit stream is perfectly capable of holding quite intelligible
    audio. It will have a broadband dynamic range of only 6 dB, but
    that is quite enough for intelligible speech and easily recognizable
    music.

    You should maybe review works such as Lipshitz and Vanderkooy's
    "Resolution below the least significant bit in audio systems with
    dither" from JAES before making such a pronouncement.
  22. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Me <secad@netspace.net.au> wrote in message news:<c8i3fp$1kpk$1@otis.netspace.net.au>...
    > Curious wrote:
    >
    > > I am looking for audio software that allows conversion of 16-bit WAVs
    > > and 8-bit WAVs to 1-bit WAVs. I have used Adobe Audition and CakeWalk
    > > Pyro. Neither of them work. CakeWalk does have a "bit-depth converter"
    > > as a FX, however, when I try to use it, I get a runtime error and
    > > Cakewalk automatically closes.
    >
    > A + Comparator FlipFlop
    > I/P -------|\ +------+ Q
    > | >----------| D Q|---+---- Bitstream out
    > B +--|/ C +--|> | |
    > | - | +------+ |
    > | | |
    > +-/\/\/\-----)-------------+
    > | R |
    > C === |
    > | Fs---+
    > 0V
    >
    > Several years ago I made a delay line to delay the audio for a walkie-talkie
    > so that the Tx VOX voice operated switch and Rx unmute operation would be
    > completed before the leading word spoken by the operator was transmitted.
    > I tried a bucket brigade device (MN3001) but had troubles with noise and
    > limitations on maximum delay.
    >
    > I tried the cct above as a 1-bit encoder (mentioned in a mag somewhere),
    > clocked at over 200KHz. I do not know if the cct has a name.
    >

    The above circuit looks to me very much like an implementation of a
    delta modulator, the RC circuit being a lossy integrator. This is the
    forunner to delta-sigma isn't it?

    Paavo Jumppanen
    Author of HarBal Harmonic Balancer
    http://www.har-bal.com
  23. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Dick Pierce wrote:

    > "Richard Crowley" <rcrowley7@xprt.net> wrote in message news:<10aola6pnbau090@corp.supernews.com>...
    >
    >>Of course true "1-bit" audio is indistinguishable from
    >>random noise.
    >>
    >
    >
    > In a word, wrong. In two words, completely wrong.
    >
    > A 1-bit stream is perfectly capable of holding quite intelligible
    > audio. It will have a broadband dynamic range of only 6 dB, but
    > that is quite enough for intelligible speech and easily recognizable
    > music.

    Please read the thread and don't try to justify balf-baked ideas. We all
    know that data coming one bit at a time can be useful: this message will
    be transmitted that way.

    According to what he asked for, the OP wants to strip all but the sign
    bit from a wave file in hopes of getting "that old movie sound".

    > You should maybe review works such as Lipshitz and Vanderkooy's
    > "Resolution below the least significant bit in audio systems with
    > dither" from JAES before making such a pronouncement.

    Why? It's irrelevant to this thread.

    Jerry
    --
    Engineering is the art of making what you want from things you can get.
    ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
  24. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Paavo Jumppanen wrote:

    > Me <secad@netspace.net.au> wrote in message news:<c8i3fp$1kpk$1@otis.netspace.net.au>...
    >
    >>Curious wrote:
    >>
    >>
    >>>I am looking for audio software that allows conversion of 16-bit WAVs
    >>>and 8-bit WAVs to 1-bit WAVs. I have used Adobe Audition and CakeWalk
    >>>Pyro. Neither of them work. CakeWalk does have a "bit-depth converter"
    >>>as a FX, however, when I try to use it, I get a runtime error and
    >>>Cakewalk automatically closes.
    >>
    >> A + Comparator FlipFlop
    >>I/P -------|\ +------+ Q
    >> | >----------| D Q|---+---- Bitstream out
    >> B +--|/ C +--|> | |
    >> | - | +------+ |
    >> | | |
    >> +-/\/\/\-----)-------------+
    >> | R |
    >> C === |
    >> | Fs---+
    >> 0V
    >>
    >>Several years ago I made a delay line to delay the audio for a walkie-talkie
    >>so that the Tx VOX voice operated switch and Rx unmute operation would be
    >>completed before the leading word spoken by the operator was transmitted.
    >>I tried a bucket brigade device (MN3001) but had troubles with noise and
    >>limitations on maximum delay.
    >>
    >>I tried the cct above as a 1-bit encoder (mentioned in a mag somewhere),
    >>clocked at over 200KHz. I do not know if the cct has a name.
    >>
    >
    >
    > The above circuit looks to me very much like an implementation of a
    > delta modulator, the RC circuit being a lossy integrator. This is the
    > forunner to delta-sigma isn't it?
    >
    > Paavo Jumppanen
    > Author of HarBal Harmonic Balancer
    > http://www.har-bal.com

    It looks to me as if the comparator will oscillate at a variable
    frequency and a duty cycle such that the average DC in its output is the
    same as the level at its input. The flip-flop merely quantizes the
    transition times. It's effectively a Class D amplifier.

    Jerry
    --
    Engineering is the art of making what you want from things you can get.
    ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
  25. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Randy Yates wrote:

    >>> Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.
    >>>
    >>
    >> On the other hand, I guess it does,
    >
    > Absolutely it does not. A PWM signal has the property that at
    > time n*T the signal begins at a high state for p*T seconds and
    > ends at a low state for (1-p)*T seconds, 0 <= p <= 1. A 1-bit
    > bitstream does not have this property.
    >
    > Many people seem to have this misconception that a delta sigma
    > bitstream is a PWM waveform. It is not - they are two different
    > animals.

    I was alluding to the resultant signal being an asymetrical bitstream, like
    a PWM signal. Though I did qualify that it bears no relationship to any
    encoded signal.

    One "1", followed by one "0", then two "1"s followed by a "0" looks PWM to
    me.

    geoff
  26. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Geoff Wood" <geoff@paf.co.nz-nospam> writes:

    > Randy Yates wrote:
    >
    > >>> Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.
    > >>>
    > >>
    > >> On the other hand, I guess it does,
    > >
    > > Absolutely it does not. A PWM signal has the property that at
    > > time n*T the signal begins at a high state for p*T seconds and
    > > ends at a low state for (1-p)*T seconds, 0 <= p <= 1. A 1-bit
    > > bitstream does not have this property.
    > >
    > > Many people seem to have this misconception that a delta sigma
    > > bitstream is a PWM waveform. It is not - they are two different
    > > animals.
    >
    > I was alluding to the resultant signal being an asymetrical bitstream, like
    > a PWM signal.

    I have no idea what you mean.

    > Though I did qualify that it bears no relationship to any
    > encoded signal.

    What is "it"?

    > One "1", followed by one "0", then two "1"s followed by a "0" looks PWM to
    > me.

    How so? It doesn't match any definition of PWM that I know of. When I talk
    about PWM, I mean, e.g., the type of signal shown in figure one of

    http://www.embedded.com/story/OEG20010821S0096

    --
    Randy Yates
    Sony Ericsson Mobile Communications
    Research Triangle Park, NC, USA
    randy.yates@sonyericsson.com, 919-472-1124
  27. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    On Fri, 21 May 2004 08:40:43 +1200, "Geoff Wood"
    <geoff@paf.co.nz-nospam> wrote:

    >Randy Yates wrote:
    >
    >>>> Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.
    >>>>
    >>>
    >>> On the other hand, I guess it does,
    >>
    >> Absolutely it does not. A PWM signal has the property that at
    >> time n*T the signal begins at a high state for p*T seconds and
    >> ends at a low state for (1-p)*T seconds, 0 <= p <= 1. A 1-bit
    >> bitstream does not have this property.
    >>
    >> Many people seem to have this misconception that a delta sigma
    >> bitstream is a PWM waveform. It is not - they are two different
    >> animals.
    >
    >I was alluding to the resultant signal being an asymetrical bitstream, like
    >a PWM signal. Though I did qualify that it bears no relationship to any
    >encoded signal.
    >
    >One "1", followed by one "0", then two "1"s followed by a "0" looks PWM to
    >me.
    >
    >geoff
    >

    Just to be sure - there are no digits in PWM - it is an analogies
    system. It is sampled, to be sure, but not digitally.

    d
    Pearce Consulting
    http://www.pearce.uk.com
  28. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Randy Yates" <randy.yates@sonyericsson.com> wrote in message
    news:xxp65aqg529.fsf@usrts005.corpusers.net...
    > "Geoff Wood" <geoff@paf.co.nz-nospam> writes:
    >
    > > Randy Yates wrote:
    > >
    > > >>> Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.
    > > >>>
    > > >>
    > > >> On the other hand, I guess it does,
    > > >
    > > > Absolutely it does not. A PWM signal has the property that at
    > > > time n*T the signal begins at a high state for p*T seconds and
    > > > ends at a low state for (1-p)*T seconds, 0 <= p <= 1. A 1-bit
    > > > bitstream does not have this property.
    > > >
    > > > Many people seem to have this misconception that a delta sigma
    > > > bitstream is a PWM waveform. It is not - they are two different
    > > > animals.
    > >
    > > I was alluding to the resultant signal being an asymetrical bitstream, like
    > > a PWM signal.
    >
    > I have no idea what you mean.
    >
    > > Though I did qualify that it bears no relationship to any
    > > encoded signal.
    >
    > What is "it"?
    >
    > > One "1", followed by one "0", then two "1"s followed by a "0" looks PWM to
    > > me.
    >
    > How so? It doesn't match any definition of PWM that I know of. When I talk
    > about PWM, I mean, e.g., the type of signal shown in figure one of
    >
    > http://www.embedded.com/story/OEG20010821S0096


    Randy,

    I don't see any reason why you couldn't create the waveforms shown in Fig. 1
    above with a 1-bit PCM waveform, assuming sufficient sample rate.

    -Jon
  29. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    On Thu, 20 May 2004 21:37:27 GMT, donald@pearce.uk.com (Don Pearce)
    wrote:

    >On Fri, 21 May 2004 08:40:43 +1200, "Geoff Wood"
    ><geoff@paf.co.nz-nospam> wrote:
    >
    >>Randy Yates wrote:
    >>
    >>>>> Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.
    >>>>>
    >>>>
    >>>> On the other hand, I guess it does,
    >>>
    >>> Absolutely it does not. A PWM signal has the property that at
    >>> time n*T the signal begins at a high state for p*T seconds and
    >>> ends at a low state for (1-p)*T seconds, 0 <= p <= 1. A 1-bit
    >>> bitstream does not have this property.
    >>>
    >>> Many people seem to have this misconception that a delta sigma
    >>> bitstream is a PWM waveform. It is not - they are two different
    >>> animals.
    >>
    >>I was alluding to the resultant signal being an asymetrical bitstream, like
    >>a PWM signal. Though I did qualify that it bears no relationship to any
    >>encoded signal.
    >>
    >>One "1", followed by one "0", then two "1"s followed by a "0" looks PWM to
    >>me.
    >>
    >>geoff
    >>
    >
    >Just to be sure - there are no digits in PWM - it is an analogies
    >system. It is sampled, to be sure, but not digitally.
    >
    >d
    >Pearce Consulting
    >http://www.pearce.uk.com

    Good old spell-checker! Make that "analogue".

    d
    Pearce Consulting
    http://www.pearce.uk.com
  30. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Jon Harris" <goldentully@hotmail.com> writes:

    > "Randy Yates" <randy.yates@sonyericsson.com> wrote in message
    > news:xxp65aqg529.fsf@usrts005.corpusers.net...
    > > "Geoff Wood" <geoff@paf.co.nz-nospam> writes:
    > >
    > > > Randy Yates wrote:
    > > >
    > > > >>> Bit-depth reducing to 1 bit does not convert PCM into a PWM signal.
    > > > >>>
    > > > >>
    > > > >> On the other hand, I guess it does,
    > > > >
    > > > > Absolutely it does not. A PWM signal has the property that at
    > > > > time n*T the signal begins at a high state for p*T seconds and
    > > > > ends at a low state for (1-p)*T seconds, 0 <= p <= 1. A 1-bit
    > > > > bitstream does not have this property.
    > > > >
    > > > > Many people seem to have this misconception that a delta sigma
    > > > > bitstream is a PWM waveform. It is not - they are two different
    > > > > animals.
    > > >
    > > > I was alluding to the resultant signal being an asymetrical bitstream, like
    > > > a PWM signal.
    > >
    > > I have no idea what you mean.
    > >
    > > > Though I did qualify that it bears no relationship to any
    > > > encoded signal.
    > >
    > > What is "it"?
    > >
    > > > One "1", followed by one "0", then two "1"s followed by a "0" looks PWM to
    > > > me.
    > >
    > > How so? It doesn't match any definition of PWM that I know of. When I talk
    > > about PWM, I mean, e.g., the type of signal shown in figure one of
    > >
    > > http://www.embedded.com/story/OEG20010821S0096
    >
    >
    > Randy,
    >
    > I don't see any reason why you couldn't create the waveforms shown in Fig. 1
    > above with a 1-bit PCM waveform, assuming sufficient sample rate.

    Jon,

    I'm not saying you can't generate PWM digitally. Yes, I agree that you can.

    I'm saying that when you requantize a multi-bit waveform to 1 bit
    (just a simple requantization, like when you requantize from 24 bits
    to 16 bits, but this time you go all the way down to 1 bit), the
    result will NOT be a PWM signal.

    I'm also saying that if you perform a delta sigma conversion of a
    multi-bit waveform to a 1-bit waveform (i.e., perform a delta sigma
    D/A conversion and examine the 1-bit bistream before it is converted
    to analog), the result will NOT be a PWM signal.
    --
    Randy Yates
    Sony Ericsson Mobile Communications
    Research Triangle Park, NC, USA
    randy.yates@sonyericsson.com, 919-472-1124
  31. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Randy Yates wrote:

    >
    > Without entering into the debate about Dick's response, let me say, Richard,
    > that I think even a 1-bit, 44.1 kHz signal will not be PURELY noise-like.

    It's been about 15 years ago but I once played around with something like
    this. Don't remember all the exact details. But, start with a good quality
    speech recording sampled at somewhere in the 10-20KHz range 8-bit signed
    integers (or maybe it was 16 bit). Then filter out most of the low frequency
    content - I think at this point most of the frequency content was in the
    1-4KHz range (sounds very tinny but still understandable)), then quantized
    down to one bit (set all but sign bit to zero) then apply a low pass filter to
    smooth out the square waves. On playback the result was surprisingly good -
    well at least as good as early sound recordings. It worked quite well for
    speech but music sounded very strange.


    > The noise will be high, but my intuition tells me you will be able to hear
    > the signal (at least one that is at a high level) in the noise. Almost
    > certainly one would be able to hear a full-scale sinewave in such noise.

    Why would the scale of the sine wave make any difference or maybe I
    misunderstand what you're saying.

    -jim


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  32. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "jim" <"N0sp"@m.sjedging@mwt.net> wrote in message
    news:40ae12c8_4@corp.newsgroups.com...
    >
    >
    > Randy Yates wrote:
    >
    > > The noise will be high, but my intuition tells me you will be able to hear
    > > the signal (at least one that is at a high level) in the noise. Almost
    > > certainly one would be able to hear a full-scale sinewave in such noise.
    >
    > Why would the scale of the sine wave make any difference or maybe I
    > misunderstand what you're saying.

    It is simply a signal-to-noise ratio issue. Much noise is added by the
    dithering/quantizing process. If the original signal is quite loud, it will
    still be recognizable above the noise floor. If it is very low level, it will
    be further buried by the noise. I did a quick experiment, and a full scale
    sinewave quanitzed with dither was easily heard. I decreased the level and
    somewhere around 20-30dB down, you really start to lose it. I was actually
    suprised by how low you could go and still make out the tone in the noise.
  33. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Robert Gault" <robert.gault@worldnet.att.net> wrote in message
    news:%Y1rc.8040$fF3.196740@bgtnsc05-news.ops.worldnet.att.net...
    > Curious wrote:
    >
    > > Randy Yates <yates@ieee.org> wrote in message
    news:<u0yfes12.fsf@ieee.org>...
    > >
    > > I want film-quality sound. By "film quality" I am referring to the
    > > crackling [resembles the sound of burning coal] and lack of clarity in
    > > the audio recordings of old B&W films which used the
    > > variable-intensity recording. I would like to preserve the frequency
    > > response of CD-audio [keep the 44.1 KHz sampling rate], however, I
    > > would like to make it mono and film-quality in terms of the artifacts
    > > mentioned above. For some reason this type of music gets my juices
    > > going.
    >
    > You may be able to get this effect with a program like CoolEditPro which
    > can merge a wide variety of distortion effects into your original sound.
    > One of the built-in distortion settings may produce the type of sound
    > you are after or you can create your own distortion.
    >
    > If you own one of these noisy films, record the sound from a section of
    > film that has mostly noise and no audio. A program like CoolEditPro can
    > extract the noise from the film section and merge it into any other
    > sound file you have.

    I think this is the right approach. This type of film noise is bursty, not like
    quantization noise. I would probably add some low level pink noise to increase
    the noise floor (easier than quantizing) and then find/create a suitable
    "crackly" sound file to mix in. This file could be looped as necessary if it
    wasn't as long as the sound track.
  34. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Richard Crowley" <rcrowley7@xprt.net> writes:

    > "TonyP" <TonyP@optus.net.com.au> wrote in message
    > news:40adc228$0$1584$afc38c87@news.optusnet.com.au...
    >>
    >> "Jerry Avins" <jya@ieee.org> wrote in message
    >> news:40ad4ecf$0$3106$61fed72c@news.rcn.com...
    >> > Dick Pierce wrote:
    >> > > "Richard Crowley" <rcrowley7@xprt.net> wrote in message
    >> news:<10aola6pnbau090@corp.supernews.com>...
    >> > >>Of course true "1-bit" audio is indistinguishable from
    >> > >>random noise.
    >> > > In a word, wrong. In two words, completely wrong.
    >> > > A 1-bit stream is perfectly capable of holding quite intelligible
    >> > > audio. It will have a broadband dynamic range of only 6 dB, but
    >> > > that is quite enough for intelligible speech and easily recognizable
    >> > > music.
    >>
    >> > Please read the thread and don't try to justify balf-baked ideas.
    >>
    >> Nothing half baked about Dicks ideas!
    >>
    >> > > You should maybe review works such as Lipshitz and Vanderkooy's
    >> > > "Resolution below the least significant bit in audio systems with
    >> > > dither" from JAES before making such a pronouncement.
    >>
    >> > Why? It's irrelevant to this thread.
    >>
    >> Dick specifically responded to Richards incorrect statement. Your post is
    >> irrelevant to that.
    >
    > My statement was not incorrect. If you go back and read
    > the OP's theory, he wanted to reduce NORMAL SAMPLING
    > RATE (i.e. 44-48K) streams to very low (i.e. 1) bit depth.
    >
    > I still maintain that a 44 or 48K by 1-bit stream is virtually
    > indistinguishable from noise.

    Without entering into the debate about Dick's response, let me say, Richard,
    that I think even a 1-bit, 44.1 kHz signal will not be PURELY noise-like.
    The noise will be high, but my intuition tells me you will be able to hear
    the signal (at least one that is at a high level) in the noise. Almost
    certainly one would be able to hear a full-scale sinewave in such noise.
    --
    % Randy Yates % "My Shangri-la has gone away, fading like
    %% Fuquay-Varina, NC % the Beatles on 'Hey Jude'"
    %%% 919-577-9882 %
    %%%% <yates@ieee.org> % 'Shangri-La', *A New World Record*, ELO
    http://home.earthlink.net/~yatescr
  35. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Jon Harris wrote:
    >

    >
    > It is simply a signal-to-noise ratio issue. Much noise is added by the
    > dithering/quantizing process. If the original signal is quite loud, it will
    > still be recognizable above the noise floor. If it is very low level, it will
    > be further buried by the noise. I did a quick experiment, and a full scale
    > sinewave quanitzed with dither was easily heard. I decreased the level and
    > somewhere around 20-30dB down, you really start to lose it. I was actually
    > suprised by how low you could go and still make out the tone in the noise.

    Well that was the point I was making in general there isn't going to be any
    significant difference between fullscale and halfscale, for example. No
    difference at all if they both have the same SNR.
    I'm also not so sure that dithering helps any when you reduce the bit depth
    below 2 bits. But maybe I just don't know the _proper_ way to dither when
    going down to 1 bit.

    -jim


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  36. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    jim <"N0sp"@m.sjedging@mwt.net> writes:
    > [...]
    > Why would the scale of the sine wave make any difference or maybe I
    > misunderstand what you're saying.

    Simple - signal-to-noise ratio. The higher the sine wave (up to
    FS clipping) the higher the signal power and the better the SNR.
    If you use dither, I would think, even with only 1 bit, you could
    hear a sine wave at lower levels than FS, but FS would almost
    definitely be audible.

    Wait - I'll try it...

    Yep, you can hear a FS sine wave just fine - boomin'! A Chopin
    piano piece came through lovely too, just with a bunch of noise.

    Here are the Matlab scripts in case you want to try this at home:

    % Modify as required
    [chopin,Fs,N] = wavread('~/wav/chopin');
    wavwrite(quantize(chopin,1,1),Fs, N, 'chopinq');

    quantize.m
    ----------
    function y = quantize(x,N,ditherBool)
    %function y = quantize(x,N,dBool)
    % FUNCTION: N-bit Dithered Quantizer
    % AUTHOR: Randy Yates
    % DATE: 05-21-2004
    % PARAMETERS:
    % x = real input vector to be quantized, in the range -1 to +1.
    % N = number of bits
    % ditherBool
    % 0 = do NOT dither
    % 1 = add TPDF dither of appropriate amplitude to x before quantizing
    % DESCRIPTION:
    % This function takes the input x and quantizes it N bits.
    % Both the input and output will be in the range -1 to +1.

    % generate dither
    d = ditherBool*(rand(size(x)) + rand(size(x)) - 1);
    x1 = x*2^(N-1) + 2^(N-1) - 1/2;
    x2 = clip(x1+d,-0.5+20*eps,2^N - 1 - 20*eps);
    x2 = round(x2);
    x3 = x2* (2/(2^N - 1)) - 1;
    y = x3;


    clip.m
    ------
    function y = clip(x, Mmin, Mmax)
    %function y = clip(x, min, max)
    % FUNCTION: clip
    % AUTHOR: Randy Yates
    % DATE: 05-21-2004
    % PARAMETERS:
    % x = real input vector to be clipped
    % Mmin = lower clip level
    % Mmax = upper clip level
    % DESCRIPTION:
    % The input will be clipped so that Mmin <= y <= Mmax

    y = min(x,Mmax);
    y = max(y,Mmin);


    --
    Randy Yates
    Sony Ericsson Mobile Communications
    Research Triangle Park, NC, USA
    randy.yates@sonyericsson.com, 919-472-1124
  37. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    jim <"N0sp"@m.sjedging@mwt.net> writes:

    > Jon Harris wrote:
    > >
    >
    > >
    > > It is simply a signal-to-noise ratio issue. Much noise is added by the
    > > dithering/quantizing process. If the original signal is quite loud, it will
    > > still be recognizable above the noise floor. If it is very low level, it will
    > > be further buried by the noise. I did a quick experiment, and a full scale
    > > sinewave quanitzed with dither was easily heard. I decreased the level and
    > > somewhere around 20-30dB down, you really start to lose it. I was actually
    > > suprised by how low you could go and still make out the tone in the noise.
    >
    > Well that was the point I was making in general there isn't going to be any
    > significant difference between fullscale and halfscale, for example. No
    > difference at all if they both have the same SNR.

    They *WON'T* have the same SNR. THAT'S THE POINT!

    > I'm also not so sure that dithering helps any when you reduce the bit depth
    > below 2 bits.

    The coarser the quantizer, the more dither helps, all the way down to 1 bit!

    > But maybe I just don't know the _proper_ way to dither when
    > going down to 1 bit.

    Same way as for N bits. The dither level for N bits is a function of N, just
    set N = 1.
    --
    Randy Yates
    Sony Ericsson Mobile Communications
    Research Triangle Park, NC, USA
    randy.yates@sonyericsson.com, 919-472-1124
  38. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Randy Yates" <randy.yates@sonyericsson.com> wrote in message
    news:xxpisepcyg7.fsf@usrts005.corpusers.net...
    > jim <"N0sp"@m.sjedging@mwt.net> writes:
    >
    > > Jon Harris wrote:
    > > >
    > > > It is simply a signal-to-noise ratio issue. Much noise is added by the
    > > > dithering/quantizing process. If the original signal is quite loud, it
    will
    > > > still be recognizable above the noise floor. If it is very low level, it
    will
    > > > be further buried by the noise. I did a quick experiment, and a full
    scale
    > > > sinewave quanitzed with dither was easily heard. I decreased the level
    and
    > > > somewhere around 20-30dB down, you really start to lose it. I was
    actually
    > > > suprised by how low you could go and still make out the tone in the noise.
    > >
    > > Well that was the point I was making in general there isn't going to be any
    > > significant difference between fullscale and halfscale, for example. No
    > > difference at all if they both have the same SNR.
    >
    > They *WON'T* have the same SNR. THAT'S THE POINT!

    Right! Halfscale with have 6dB less SNR than full-scale.
  39. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Thanks for posting Randy. FYI, I accomplished a similar thing using CoolEdit.
    I first reduced the gain by 90dB, making sure the Dither option was checked in
    preferences. Then I normalized the file to boost the level to where I could
    easily hear it. It's not strictly 1-bit, because you end up with 3 levels
    (-full scale, zero, +full scale), so it's more than 1.5 bit. But it
    illustrates the general effect quite nicely.

    "Randy Yates" <randy.yates@sonyericsson.com> wrote in message
    news:xxpn041cyuf.fsf@usrts005.corpusers.net...
    > Wait - I'll try it...
    >
    > Yep, you can hear a FS sine wave just fine - boomin'! A Chopin
    > piano piece came through lovely too, just with a bunch of noise.
    >
    > Here are the Matlab scripts in case you want to try this at home:
    >
    > % Modify as required
    > [chopin,Fs,N] = wavread('~/wav/chopin');
    > wavwrite(quantize(chopin,1,1),Fs, N, 'chopinq');
    >
    > quantize.m
    > ----------
    > function y = quantize(x,N,ditherBool)
    > %function y = quantize(x,N,dBool)
    > % FUNCTION: N-bit Dithered Quantizer
    > % AUTHOR: Randy Yates
    > % DATE: 05-21-2004
    > % PARAMETERS:
    > % x = real input vector to be quantized, in the range -1 to +1.
    > % N = number of bits
    > % ditherBool
    > % 0 = do NOT dither
    > % 1 = add TPDF dither of appropriate amplitude to x before quantizing
    > % DESCRIPTION:
    > % This function takes the input x and quantizes it N bits.
    > % Both the input and output will be in the range -1 to +1.
    >
    > % generate dither
    > d = ditherBool*(rand(size(x)) + rand(size(x)) - 1);
    > x1 = x*2^(N-1) + 2^(N-1) - 1/2;
    > x2 = clip(x1+d,-0.5+20*eps,2^N - 1 - 20*eps);
    > x2 = round(x2);
    > x3 = x2* (2/(2^N - 1)) - 1;
    > y = x3;
    >
    >
    > clip.m
    > ------
    > function y = clip(x, Mmin, Mmax)
    > %function y = clip(x, min, max)
    > % FUNCTION: clip
    > % AUTHOR: Randy Yates
    > % DATE: 05-21-2004
    > % PARAMETERS:
    > % x = real input vector to be clipped
    > % Mmin = lower clip level
    > % Mmax = upper clip level
    > % DESCRIPTION:
    > % The input will be clipped so that Mmin <= y <= Mmax
    >
    > y = min(x,Mmax);
    > y = max(y,Mmin);
  40. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "Jerry Avins" <jya@ieee.org> wrote in message
    news:40ad4ecf$0$3106$61fed72c@news.rcn.com...
    > Dick Pierce wrote:
    > > "Richard Crowley" <rcrowley7@xprt.net> wrote in message
    news:<10aola6pnbau090@corp.supernews.com>...
    > >>Of course true "1-bit" audio is indistinguishable from
    > >>random noise.
    > > In a word, wrong. In two words, completely wrong.
    > > A 1-bit stream is perfectly capable of holding quite intelligible
    > > audio. It will have a broadband dynamic range of only 6 dB, but
    > > that is quite enough for intelligible speech and easily recognizable
    > > music.

    > Please read the thread and don't try to justify balf-baked ideas.

    Nothing half baked about Dicks ideas!

    > > You should maybe review works such as Lipshitz and Vanderkooy's
    > > "Resolution below the least significant bit in audio systems with
    > > dither" from JAES before making such a pronouncement.

    > Why? It's irrelevant to this thread.

    Dick specifically responded to Richards incorrect statement. Your post is
    irrelevant to that.

    TonyP.
  41. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "TonyP" <TonyP@optus.net.com.au> wrote in message
    news:40adc228$0$1584$afc38c87@news.optusnet.com.au...
    >
    > "Jerry Avins" <jya@ieee.org> wrote in message
    > news:40ad4ecf$0$3106$61fed72c@news.rcn.com...
    > > Dick Pierce wrote:
    > > > "Richard Crowley" <rcrowley7@xprt.net> wrote in message
    > news:<10aola6pnbau090@corp.supernews.com>...
    > > >>Of course true "1-bit" audio is indistinguishable from
    > > >>random noise.
    > > > In a word, wrong. In two words, completely wrong.
    > > > A 1-bit stream is perfectly capable of holding quite intelligible
    > > > audio. It will have a broadband dynamic range of only 6 dB, but
    > > > that is quite enough for intelligible speech and easily recognizable
    > > > music.
    >
    > > Please read the thread and don't try to justify balf-baked ideas.
    >
    > Nothing half baked about Dicks ideas!
    >
    > > > You should maybe review works such as Lipshitz and Vanderkooy's
    > > > "Resolution below the least significant bit in audio systems with
    > > > dither" from JAES before making such a pronouncement.
    >
    > > Why? It's irrelevant to this thread.
    >
    > Dick specifically responded to Richards incorrect statement. Your post is
    > irrelevant to that.

    My statement was not incorrect. If you go back and read
    the OP's theory, he wanted to reduce NORMAL SAMPLING
    RATE (i.e. 44-48K) streams to very low (i.e. 1) bit depth.

    I still maintain that a 44 or 48K by 1-bit stream is virtually
    indistinguishable from noise. Most certainly, if you greatly
    increase the sampling rate (as any 1-bit D/A converter does)
    you restore "fidelity". But this was NOT the OPs theory.

    One of the characeristics of Usenet it the great tendency to go
    off and discuss semi-related theory even when it has no impact
    on the original question. On one side, this is a benefit for those
    lurking to learn some bits of theory. OTOH, it leads some to
    completely falacious associations between uncorrelated theory
    and the original issue.
  42. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    TonyP wrote:

    > "Jerry Avins" <jya@ieee.org> wrote in message
    > news:40ad4ecf$0$3106$61fed72c@news.rcn.com...
    >
    >>Dick Pierce wrote:
    >>
    >>>"Richard Crowley" <rcrowley7@xprt.net> wrote in message
    >
    > news:<10aola6pnbau090@corp.supernews.com>...
    >
    >>>>Of course true "1-bit" audio is indistinguishable from
    >>>>random noise.
    >>>
    >>>In a word, wrong. In two words, completely wrong.
    >>>A 1-bit stream is perfectly capable of holding quite intelligible
    >>>audio. It will have a broadband dynamic range of only 6 dB, but
    >>>that is quite enough for intelligible speech and easily recognizable
    >>>music.
    >
    >
    >>Please read the thread and don't try to justify balf-baked ideas.
    >
    >
    > Nothing half baked about Dicks ideas!

    As far as facts go, Dick is right on the mark. Nothing half baked there.
    The notion of making sound with just the sign bit of a sampled waveform
    and expecting it to sound like an old movie sound track: that's half baked.

    >>>You should maybe review works such as Lipshitz and Vanderkooy's
    >>>"Resolution below the least significant bit in audio systems with
    >>>dither" from JAES before making such a pronouncement.
    >
    >
    >>Why? It's irrelevant to this thread.
    >
    >
    > Dick specifically responded to Richards incorrect statement. Your post is
    > irrelevant to that.

    He responded with IMHO irrelevant examples useful of 1-bit signals. (He
    could have added delta modulation and bit-serial data protocols.)

    > TonyP.
    >
    >

    Jerry
    --
    Engineering is the art of making what you want from things you can get.
    ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
  43. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Richard Crowley wrote:

    ...


    > My statement was not incorrect. If you go back and read
    > the OP's theory, he wanted to reduce NORMAL SAMPLING
    > RATE (i.e. 44-48K) streams to very low (i.e. 1) bit depth.
    >
    > I still maintain that a 44 or 48K by 1-bit stream is virtually
    > indistinguishable from noise. Most certainly, if you greatly
    > increase the sampling rate (as any 1-bit D/A converter does)
    > you restore "fidelity". But this was NOT the OPs theory.

    Reducing the bit depth to one is the same as extracting the sign bit,
    which in turn is the same as clipping with very high gain. One can hear
    what this sounds like by running an audio signal into a comparator with
    bi-polar output. (Make one from an op-amp.) It isn't pretty; you can
    hear some beat, but no recognizable tune. Cacophony? Yes. Noise? Well... no.

    > One of the characeristics of Usenet it the great tendency to go
    > off and discuss semi-related theory even when it has no impact
    > on the original question. On one side, this is a benefit for those
    > lurking to learn some bits of theory. OTOH, it leads some to
    > completely falacious associations between uncorrelated theory
    > and the original issue.

    Right on. You said what I meant.

    Jerry
    --
    Engineering is the art of making what you want from things you can get.
    ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
  44. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Richard Crowley <rcrowley7@xprt.net> wrote:
    > I still maintain that a 44 or 48K by 1-bit stream is virtually
    > indistinguishable from noise.

    Have you actually tried it? I have (in Matlab) and it's actually quite
    surprising just how intelligible the result still is (not that you'd mistake
    it for 'hi fi' in your lifetime...).

    ---Joel Kolstad
  45. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Resuming my self-imposed role as "playground cop", here's the deal as I see it:

    There are actually 2 different issues up for debate in this thread:
    1. Does quantizing to 1-bit audio give that "old time movie sound"? I think we
    all agree the answer is no!
    2. Does quantizing to 1-bit audio yield pure noise or can you still recognize
    the audio? From my experiments, the answer is clearly that when _dithered
    properly_ and the original signal was large enough, you can still definitely
    recognize the audio, though it is buried in noise. It sounds somewhat like
    recording something at a very low level on a cassette deck and then really
    cranking up the volume on playback. The tape hiss is very audible, but you can
    still make out the audio, noisy as it is.

    Most of the confusion seems to be due to not recognizing the 2 separate
    questions. IMHO.
    -Jon

    "Jerry Avins" <jya@ieee.org> wrote in message
    news:40ae15cb$0$3107$61fed72c@news.rcn.com...
    > TonyP wrote:
    >
    > > "Jerry Avins" <jya@ieee.org> wrote in message
    > > news:40ad4ecf$0$3106$61fed72c@news.rcn.com...
    > >
    > >>Dick Pierce wrote:
    > >>
    > >>>"Richard Crowley" <rcrowley7@xprt.net> wrote in message
    > >
    > > news:<10aola6pnbau090@corp.supernews.com>...
    > >
    > >>>>Of course true "1-bit" audio is indistinguishable from
    > >>>>random noise.
    > >>>
    > >>>In a word, wrong. In two words, completely wrong.
    > >>>A 1-bit stream is perfectly capable of holding quite intelligible
    > >>>audio. It will have a broadband dynamic range of only 6 dB, but
    > >>>that is quite enough for intelligible speech and easily recognizable
    > >>>music.
    > >
    > >
    > >>Please read the thread and don't try to justify balf-baked ideas.
    > >
    > >
    > > Nothing half baked about Dicks ideas!
    >
    > As far as facts go, Dick is right on the mark. Nothing half baked there.
    > The notion of making sound with just the sign bit of a sampled waveform
    > and expecting it to sound like an old movie sound track: that's half baked.
    >
    > >>>You should maybe review works such as Lipshitz and Vanderkooy's
    > >>>"Resolution below the least significant bit in audio systems with
    > >>>dither" from JAES before making such a pronouncement.
    > >
    > >
    > >>Why? It's irrelevant to this thread.
    > >
    > >
    > > Dick specifically responded to Richards incorrect statement. Your post is
    > > irrelevant to that.
    >
    > He responded with IMHO irrelevant examples useful of 1-bit signals. (He
    > could have added delta modulation and bit-serial data protocols.)
  46. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Jerry Avins <jya@ieee.org> writes:

    > Richard Crowley wrote:
    >
    > ...
    >
    >
    > > My statement was not incorrect. If you go back and read
    > > the OP's theory, he wanted to reduce NORMAL SAMPLING
    > > RATE (i.e. 44-48K) streams to very low (i.e. 1) bit depth.
    > > I still maintain that a 44 or 48K by 1-bit stream is virtually
    >
    > > indistinguishable from noise. Most certainly, if you greatly
    > > increase the sampling rate (as any 1-bit D/A converter does)
    > > you restore "fidelity". But this was NOT the OPs theory.
    >
    > Reducing the bit depth to one is the same as extracting the sign bit,
    > which in turn is the same as clipping with very high gain. One can
    > hear what this sounds like by running an audio signal into a
    > comparator with bi-polar output. (Make one from an op-amp.) It isn't
    > pretty; you can hear some beat, but no recognizable tune. Cacophony?
    > Yes. Noise? Well... no.

    I was presuming the appropriate amount of dither was used in the
    requantization. Otherwise it would be much much worse sounding,
    and, I agree, non-noise-like.
    --
    Randy Yates
    Sony Ericsson Mobile Communications
    Research Triangle Park, NC, USA
    randy.yates@sonyericsson.com, 919-472-1124
  47. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    Jon Harris wrote:

    > >
    > > They *WON'T* have the same SNR. THAT'S THE POINT!
    >
    > Right! Halfscale with have 6dB less SNR than full-scale.

    Not necessarily. Its perfectly possible that the same bit pattern will result
    from both input sequences or an even cleaner one from the half scale if the
    full scale contains more noise. I guess your assuming that quantization error
    is the only source of noise in the original signal. Even if that were true the
    difference in error from half-scale to full scale is insignificant compared to
    the amount of error added when going down to one bit - so few bits will end up
    different that you won't be able to hear the difference. If I understand
    Randy's dither algo the amount of dither added decreases proportional to the
    final bit depth. Again that means when you get down to one bit so few bits
    will be changed you won't be able to hear the difference.
    By the way, I'm surprised that no one picked up on the fact that coal doesn't
    crackle when it burns :-}

    -jim


    -----= Posted via Newsfeeds.Com, Uncensored Usenet News =-----
    http://www.newsfeeds.com - The #1 Newsgroup Service in the World!
    -----== Over 100,000 Newsgroups - 19 Different Servers! =-----
  48. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    "jim" <"N0sp"@m.sjedging@mwt.net> wrote in message
    news:40ae9845_6@corp.newsgroups.com...
    >
    >
    > Jon Harris wrote:
    >
    > > >
    > > > They *WON'T* have the same SNR. THAT'S THE POINT!
    > >
    > > Right! Halfscale with have 6dB less SNR than full-scale.
    >
    > Not necessarily. Its perfectly possible that the same bit pattern will result
    > from both input sequences or an even cleaner one from the half scale if the
    > full scale contains more noise.

    I thought we were talking about the difference between full-scale and half-scale
    sine waves?

    > I guess your assuming that quantization error
    > is the only source of noise in the original signal. Even if that were true the
    > difference in error from half-scale to full scale is insignificant compared to
    > the amount of error added when going down to one bit - so few bits will end up
    > different that you won't be able to hear the difference. If I understand
    > Randy's dither algo the amount of dither added decreases proportional to the
    > final bit depth.

    I think you have that backwards. The fewer bits, the more dither you need, so
    dither is inversely proportional to final bit depth.

    > Again that means when you get down to one bit so few bits
    > will be changed you won't be able to hear the difference.

    I think you are a bit confused about the way dithered quantization works. Given
    a clean input signal, properly dithered quantization to N bits adds noise based
    on N alone--it is NOT depended on the signal. Now if you quantize a signal
    identical to the first one in every way except 6dB quieter again to N bits
    (assuming the original signal has >> bits than N) you will have the same amount
    of quantization noise, but since the original signal was 6dB softer, you have
    6dB worse SNR. It really works, I've tried it!

    I'll use my analog cassette tape analogy one more time. Imagine recording a
    very low level signal to a cassette tape. Then boost the gain on playback, say
    60dB. You can still hear the recorded signal, but there is plenty of tape hiss.
    Now imagine doing the same thing again, except with the original signal 6dB
    softer. Boost by the same 60dB and the tape hiss is still at the same level,
    but the original signal is 6dB softer, hence 6dB worse SNR. Dithered
    quantization works THE SAME WAY! The dither (noise) creates a noise floor just
    like tape hiss. (The only difference is that the frequency response of the tape
    noise may be different than the dither noise. In fact, the digital designer can
    choose the sound of the noise floor using noise shaping.)

    BTW, that's one of the big breakthroughs about dither--it makes digital sound
    like analog!
  49. Archived from groups: rec.audio.tech,comp.dsp (More info?)

    On 2004-05-21, Jon Harris <goldentully@hotmail.com> wrote:
    > "jim" <"N0sp"@m.sjedging@mwt.net> wrote in message
    > news:40ae12c8_4@corp.newsgroups.com...
    >>
    >>
    >> Randy Yates wrote:
    >>
    >> > The noise will be high, but my intuition tells me you will be able to hear
    >> > the signal (at least one that is at a high level) in the noise. Almost
    >> > certainly one would be able to hear a full-scale sinewave in such noise.
    >>
    >> Why would the scale of the sine wave make any difference or maybe I
    >> misunderstand what you're saying.
    >
    > It is simply a signal-to-noise ratio issue. Much noise is added by the
    > dithering/quantizing process. If the original signal is quite loud, it will
    > still be recognizable above the noise floor. If it is very low level, it will
    > be further buried by the noise. I did a quick experiment, and a full scale
    > sinewave quanitzed with dither was easily heard. I decreased the level and
    > somewhere around 20-30dB down, you really start to lose it. I was actually
    > suprised by how low you could go and still make out the tone in the noise.

    Here is an experiment I did.

    There are 16 repetitions of the well-known 909 kick drum, mixing triangular noise
    in varying amounts and quantizing to 1 bit. Each hit of the drum doubles the
    noise. It's really interesting to look at with a wave editor, as well as to
    listen. At the low-noise version, you hear a "mean" kick sound. At the noisy end,
    you hear more dynamic range and more noise. I personally like it best with
    about 3/4 of an LSB added. BTW, the subjective volume seems quieter when
    more noise is added:

    http://www.gweep.net/~shifty/audio/1bit/reduction/re02.wav

    best listened to in loop mode!! :)


    --
    different MP3 every day! http://gweep.net/~shifty/snackmaster
    . . . . . . . . ... . . . . . .
    "Maybe if you ever picked up a goddamn keyboard | Niente
    and compiler, you'd know yourself." -Matthew 7:1 | shifty@gweep.net
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