View source for Videos/Digital Show and Tell

<small>''Wiki edition''</small>
[[Image:dsat_001.jpg|400px|right]]

Continuing in the "firehose" tradition of [[Videos/A_Digital_Media_Primer_For_Geeks|Episode 01]], Xiph.Org's second video on digital media explores multiple facets of digital audio signals and how they ''really'' behave in the real world.

Demonstrations of sampling, quantization, bit-depth, and dither explore digital audio behavior on real audio equipment using both modern digital analysis and vintage analog bench equipment, just in case we can't trust those newfangled digital gizmos. You can download the source code for each demo and try it all for yourself!
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[[Image:Xiph_ep02_test.png|400px|right]]

Hi, I'm Monty Montgomery from [http://www.redhat.com/ Red Hat] and [http://xiph.org/ Xiph.Org].

A few months ago, I wrote
[http://people.xiph.org/~xiphmont/demo/neil-young.html an article on digital audio and why 24bit/192kHz music downloads don't make sense].
In the article, I
mentioned--almost in passing--that a digital waveform is
[http://people.xiph.org/~xiphmont/demo/neil-young.html#toc_sfam not a stairstep],
and you certainly don't get a stairstep when you
[[WikiPedia:Digital-to-analog_converter|convert from digital back to analog]].

Of everything in the entire article, '''that''' was the number one thing
people wrote about. In fact, more than half the mail I got was questions and
comments about basic digital signal behavior.  Since there's interest, let's
take a little time to play with some ''simple'' digital signals.

==Veritas ex machina==
[[Image:Dsat_002.jpg|200px|right]]
[[Image:Dsat_003.jpg|200px|right]]
[[Image:Dsat_004.jpg|200px|right]]
[[Image:Dsat_005.jpg|200px|right]]

Pretend for a moment that we have no idea how digital signals really
behave. In that case it doesn't make sense for us to use digital test
equipment either.  Fortunately for this exercise, there's still plenty
of working analog lab equipment out there.

First up, we need a [[WikiPedia:Function_generator|signal generator]] to provide us with analog input
signals--in this case, an
[http://www.home.agilent.com/en/pd-3325A%3Aepsg%3Apro-pn-3325A/synthesizer-function-generator?pm=PL&nid=-536900197.536896863&cc=SE&lc=swe HP3325]
from 1978.  It's still a pretty good
generator, so if you don't mind the size, the weight, the power
consumption, and the noisy fan, you can find them on eBay... occasionally
for only slightly more than you'll pay for shipping.

Next, we'll observe our analog waveforms on [[WikiPedia:Oscilloscope_types#Cathode-ray_oscilloscope_.28CRO.29|analog oscilloscopes]],
like this Tektronix 2246 from the mid-90s, one of the last and very best analog scopes ever made. Every home lab should have one.

...and finally inspect the [[WikiPedia:Spectral_density#Electrical_engineering|frequency spectrum]] of our signals using an
[[WikiPedia:Spectrum_analyzer#Swept-tuned|analog spectrum analyzer]], this
[http://www.home.agilent.com/en/pd-3585A%3Aepsg%3Apro-pn-3585A/spectrum-analyzer-high-perf-20hz-40mhz?pm=PL&nid=-536900197.536897319&cc=SE&lc=swe HP3585]
from the same product line as
the signal generator.  Like the other equipment here it has
[http://www.hp9845.net/9845/hardware/processors/ a rudimentary and hilariously large microcontroller],
but the signal path
from input to what you see on the screen is completely analog.

All of this equipment is vintage, but aside from its raw tonnage, the specs are still quite good.

At the moment, we have our signal generator set to output a nice 1 [[WikiPedia:Hertz#SI_multiples|kHz]]
sine wave at one [[WikiPedia:Volt|Volt]] [[WikiPedia:Amplitude#Root_mean_square_amplitude|RMS]].
We see the sine wave on the oscilloscope, can verify that it is indeed
1 kHz at 1 Volt RMS, which is 2.8 Volts
[[WikiPedia:Amplitude#Peak-to-peak_amplitude|peak-to-peak]],
and that matches the
measurement on the spectrum analyzer as well.

The analyzer also shows some low-level [[WikiPedia:White_noise|white noise]]
and just a bit of [[WikiPedia:Harmonic_distortion#Harmonic_distortion|harmonic distortion]],
with the highest peak about 70[[WikiPedia:Decibel|dB]] or so below
[[WikiPedia:Fundamental_frequency|the fundamental]].
Now, this doesn't matter at all in our demos, but I
wanted to point it out now just in case you didn't notice it until
later.

Now, we drop digital sampling in the middle.

For the conversion, we'll use a boring, consumer-grade, eMagic USB1
audio device.  It's also more than ten years old at this point, and it's
getting obsolete.

A recent converter can easily have an order of magnitude better specs.
[[WikiPedia:Reconstruction_filter#Sampled_data_reconstruction_filters|Flatness]],
[[WikiPedia:Analog-to-digital_converter#Non-linearity|linearity]],
[[WikiPedia:Jitter#Sampling_jitter|jitter]],
[[WikiPedia:Noise_floor|noise behavior]],
[[WikiPedia:Digital-to-analog_converter#DAC_performance|everything]]...
you may not
have noticed.  Just because we can measure an improvement doesn't
mean we can hear it, and even these old consumer boxes were already at
the edge of ideal transparency.

The eMagic connects to my ThinkPad, which displays a digital
waveform and spectrum for comparison, then the ThinkPad
sends the digital signal right back out to the eMagic for
re-conversion to analog and observation on the output scopes.

Input to output, left to right.
<br style="clear:both;"/>

==Stairsteps==
[[Image:Dsat 006.jpg|360px|right]]
[[Image:Dsat 007.png|360px|right]]
OK, it's go time. We begin by converting an analog signal to digital and
then right back to analog again with no other steps.

The signal generator is set to produce a 1kHz sine wave just like
before.

We can see our analog sine wave on our input-side oscilloscope.

We digitize our signal to
[[Videos/A_Digital_Media_Primer_For_Geeks#Raw_.28digital_audio.29_meat|16 bit PCM at 44.1kHz]],
same as on a CD.
The spectrum of the digitized signal matches what we saw earlier

and what we see now on the analog spectrum analyzer, aside from its 
[[WikiPedia:High_impedance|high-impedance input]] being just a smidge noisier.

For now, the waveform display shows our digitized sine wave as a
stairstep pattern, one step for each sample.

And when we look at the output signal that's been converted
from digital back to analog, we see...

It's exactly like the original sine wave.  No stairsteps.

OK, 1 kHz is still a fairly low frequency, maybe the stairsteps are just
hard to see or they're being smoothed away.  Fair enough. Let's choose
a higher frequency, something close to [[WikiPedia:Nyquist_frequency|Nyquist]], say 15kHz.

Now the sine wave is represented by less than three samples per cycle, and...

the digital waveform looks pretty awful.  Well, looks
can be deceiving. The analog output...

is still a perfect sine wave, exactly like the original.

Let's keep going up.

Let's see if I can do this without blocking any cameras.

16kHz.... 17kHz... 18kHz... 19kHz... 

20kHz.  Welcome to the upper limits of human hearing. The output
waveform is still perfect. No jagged edges, no dropoff, no stairsteps.

So where'd the stairsteps go? Don't answer, it's a trick question.
They were never there.

Drawing a digital waveform as a stairstep... was wrong to begin with.

Why? A stairstep is a continuous-time function.  It's jagged, and it's
piecewise, but it has a defined value at every point in time.

A sampled signal is entirely different. It's discrete-time; it's only
got a value right at each instantaneous sample point and it's
undefined, there is no value at all, everywhere between.  A
discrete-time signal is properly drawn as a lollipop graph.

The continuous, analog counterpart of a digital signal passes
smoothly through each sample point, and that's just as true for high
frequencies as it is for low.

Now, the interesting and not at all obvious bit is: [[WikiPedia:Nyquist%E2%80%93Shannon_sampling_theorem|there's only one
bandlimited signal that passes exactly through each sample point]]. It's
a unique solution. So if you sample a bandlimited signal and then
convert it back, the original input is also the only possible output.

[[Image:Dsat 008.png|360px|right]]

And before you say, "oh, I can draw a different signal that passes
through those points", well, yes you can, but if it differs even
minutely from the original, it includes frequency content at or beyond
Nyquist, breaks the bandlimiting requirement and isn't a valid
solution.

So how did everyone get confused and start thinking of digital signals
as stairsteps? I can think of two good reasons.

First: it's easy enough to convert a sampled signal to a true stairstep. Just
extend each sample value forward until the next sample period.  This is
called a [[WikiPedia:Zero-order hold|zero-order hold]], and it's an important part of how some
digital-to-analog converters work, especially the simplest ones.

So, anyone who looks up [[WikiPedia:Digital-to-analog_converter#Practical_operation|digital-to-analog converter or
digital-to-analog conversion]] is probably going to see a diagram of a
stairstep waveform somewhere, but that's not a finished conversion,
and it's not the signal that comes out.

Second, and this is probably the more likely reason, engineers who
supposedly know better, like me, draw stairsteps even though they're
technically wrong. It's a sort of like a one-dimensional version of
[[WikiPedia:MacPaint#Development|fat bits in an image editor]].

Pixels aren't squares either, they're samples of a 2-dimensional
function space and so they're also, conceptually, infinitely small
points. Practically, it's a real pain in the ass to see or manipulate
infinitely small anything, so big squares it is.  Digital stairstep
drawings are exactly the same thing.

It's just a convenient drawing. The stairsteps aren't really there.

==Bit-depth==
[[Image:Dsat_009.jpg|360px|right]]
[[Image:Dsat_010.jpg|260px|right]]

When we convert a digital signal back to analog, the result is
''also'' smooth regardless of the [[WikiPedia:Audio_bit_depth|bit depth]].  24 bits or 16 bits...
or 8 bits...  it doesn't matter.

So does that mean that the digital bit depth makes no difference at
all? Of course not.

Channel 2 here is the same sine wave input, but we quantize with
[[WikiPedia:Dither|dither]] down to 8 bits.

On the scope, we still see a nice
smooth sine wave on channel 2. Look very close, and you'll also see a
bit more noise.  That's a clue.

If we look at the spectrum of the signal... aha!  Our sine wave is
still there unaffected, but the noise level of the 8-bit signal on
the second channel is much higher!

And that's the difference the number of bits makes.  That's it!

When we digitize a signal, first we sample it. The
sampling step is perfect; it loses nothing. But then we [[WikiPedia:Quantization_(sound_processing)|quantize]] it,
and [[WikiPedia:Quantization_error|quantization adds noise]].

The number of bits determines how much noise and so the level of the
noise floor.

What does this dithered quantization noise sound like?  Let's listen
to our 8-bit sine wave.

That may have been hard to hear anything but the tone.  Let's listen
to just the noise after we notch out the sine wave and then bring the
gain up a bit because the noise is quiet.

Those of you who have used analog recording equipment may have just
thought to yourselves, "My goodness! That sounds like tape hiss!"
Well, it doesn't just sound like tape hiss, it acts like it too, and
if we use a [[WikiPedia:Dither#Different_types|gaussian dither]] then it's
[[WikiPedia:Central_limit_theorem|mathematically equivalent]] in every way. It ''is'' tape hiss.

Intuitively, that means that we can measure tape hiss and thus the noise floor
of [[WikiPedia:Magnetic_tape_sound_recording|magnetic audio tape]]
in [[WikiPedia:Shannon–Hartley_theorem#Examples|bits instead of decibels]], in order to put things in a
digital perspective.  [[WikiPedia:Compact cassettes|Compact cassettes]] (for those of you who are old enough to remember them) could reach as
deep as 9 bits in perfect conditions, though 5 to 6 bits was
more typical, especially if it was a recording made on a
[[WikiPedia:Cassette_deck|tape deck]]. That's right... your mix tapes were only about 6 bits
deep... if you were lucky!

The very best professional [[WikiPedia:Reel-to-reel_audio_tape_recording|open reel tape]] used in studios could barely
hit...  any guesses? 13 bits ''with'' [[WikiPedia:Reel-to-reel_audio_tape_recording#Noise_reduction|advanced noise reduction]].  And
that's why seeing '[[WikiPedia:SPARS_code|D D D]]' on a [[WikiPedia:Compact_disk|Compact Disc]] used to be such a big,
high-end deal.

==Dither==
[[Image:Dsat_011.png|360px|right]]

I keep saying that I'm quantizing with [[Wikipedia:dither|dither]], so what is dither
exactly and, more importantly, what does it do?

The simple way to quantize a signal is to choose the digital
amplitude value closest to the original analog amplitude.  [[WikiPedia:Rounding|Obvious]],
right?  Unfortunately, the exact noise you get from this simple
quantization scheme depends somewhat on the input signal,

so we may get noise that's inconsistent, or causes distortion, or is
undesirable in some other way.
<br style="clear:both;"/>
<center><div style="background-color:#DDDDFF;border-color:#CCCCDD;border-style:solid;width:80%;padding:0 1em 1em 1em;text-align:left;">
'''Going deeper…'''
*Cameron Nicklaus Christou's thesis [http://uwspace.uwaterloo.ca/bitstream/10012/3867/1/thesis.pdf Optimal Dither and Noise Shaping in Image Processing] provides an ''excellent'' explanation of dither and noise shaping.
</div></center>

Dither is specially-constructed noise that substitutes for the noise
produced by simple quantization. Dither doesn't [[WikiPedia:Sound_masking|drown out or mask]]
quantization noise, it actually replaces it with noise characteristics
of our choosing that aren't influenced by the input.

Let's ''watch'' what dither does.  The signal generator has too much noise for this test so we'll produce a mathematically perfect sine wave with the ThinkPad and quantize it to 8 bits with dithering.

We see a nice sine wave on the waveform display and output scope and, once the analog spectrum analyzer catches up...
a clean frequency peak with a uniform noise floor on both spectral displays
just like before. Again, this is with dither.

Now I turn dithering off.

The quantization noise, that dither had spread out into a nice, flat noise
floor, piles up into harmonic distortion peaks.  The noise floor is
lower, but the level of distortion becomes nonzero, and the distortion
peaks sit higher than the dithering noise did.

At 8 bits this effect is exaggerated. At 16 bits,
even without dither, harmonic distortion is going to be so low as to
be completely inaudible.

Still, we can use dither to eliminate it completely if we so choose.

Turning the dither off again for a moment, you'll notice that the
absolute level of distortion from undithered quantization stays
approximately constant regardless of the input amplitude.
But when the signal level drops below a half a bit, everything
quantizes to zero.

In a sense, everything quantizing to zero is just 100% distortion!
Dither eliminates this distortion too. We reenable dither
and ... there's our signal back at 1/4 bit, with our nice flat noise floor.

The noise floor doesn't have to be flat.  Dither is noise of our
choosing, so let's choose a noise as [http://www.acoustics.salford.ac.uk/res/cox/sound_quality/?content=subjective inoffensive] and
[[WikiPedia:Absolute_threshold_of_hearing|difficult to notice]]
as possible.

Our hearing is most sensitive in the midrange from 2kHz to 4kHz,
so that's where background noise is going to be the most obvious.
We can [[WikiPedia:Noise_shaping|shape dithering noise]] away from sensitive frequencies to where
hearing is less sensitive, usually the highest frequencies.

16-bit dithering noise is normally much too quiet to hear at all, but
let's listen to our noise shaping example, again with the gain
brought way up...

Lastly, dithered quantization noise ''is'' higher [[WikiPedia:power|Sound_power]] overall
than undithered quantization noise even when it sounds quieter, and
you can see that on a [[WikiPedia:VU_meter|VU meter]] during passages of near-silence.  But
dither isn't only an on or off choice. We can reduce the dither's
power to balance less noise against a bit of distortion to minimize
the overall effect.

We'll also [[WikiPedia:Amplitude_modulation|modulate the input signal]] like this to show how a varying input affects the quantization noise.  At
full dithering power, the noise is uniform, constant, and featureless
just like we expect:

As we reduce the dither's power, the input increasingly
affects the amplitude and the character of the quantization noise.
Shaped dither behaves similarly, but noise shaping lends one more nice
advantage.  To make a long story short, it can use a somewhat lower
dither power before the input has as much effect on the output.

Despite all the time I just spent on dither, we're talking about
differences that start 100 decibels and more below [[WikiPedia:Full_scale|full scale]].  Maybe
if the CD had been
[http://www.research.philips.com/technologies/projects/cd/index.html 14 bits as originally designed],
dither ''might'' be
more important.  Maybe.  At 16 bits, really, it's mostly a wash.  You
can think of dither as an insurance policy that gives several extra
decibels of dynamic range just in case. The simple fact is, though, no
one ever ruined a great recording by not dithering the final master.

==Bandlimitation and timing==
[[image:Dsat_013.jpg|360px|right]]
[[image:Dsat_014.gif|360px|right]]

We've been using [[WikiPedia:Sine_wave|sine waves]]. They're the obvious choice when what we
want to see is a system's behavior at a given isolated frequency.  Now
let's look at something a bit more complex.  What should we expect to
happen when I change the input to a [[WikiPedia:Square_wave|square wave]]...

The input scope confirms our 1kHz square wave.  The output scope shows..


Exactly what it should.
...
What is a square wave really?  


Well, we can say it's a waveform that's some positive value for half a cycle and then transitions instantaneously to a negative value for the other half.

:<math>
\ x(t) = \begin{cases} 1, & |t| < T_1 \\ 0, & T_1 < |t| \leq {1 \over 2}T \end{cases}
</math>

But that doesn't really tell us anything useful about how that input becomes this output.

Then we remember that any waveform is also [[WikiPedia:Fourier_series|the sum of discrete frequencies]],
and a square wave is particularly simple sum: a fundamental and an
infinite series of [[WikiPedia:Even_and_odd_functions#Harmonics|odd harmonics]].  Sum them all up, you get a
square wave.

:<math>\begin{align}
x_{\mathrm{square}}(t) & {} = \frac{4}{\pi}\sin(\omega t) + \frac{4}{3\pi}\sin(3\omega t) + \frac{4}{5\pi}\sin(5\omega t) + \frac{4}{7\pi}\sin(7\omega t) + \cdots 
\end{align}</math>

At first glance, that doesn't seem very useful either. You have to sum up an infinite number of harmonics to get the answer.  ''Ah'', but we don't have an infinite number of harmonics.

We're using a quite sharp [[WikiPedia:Low-pass_filter|anti-aliasing filter]] that cuts off right
above 20kHz, so our signal is [[WikiPedia:Bandlimiting|bandlimited]], which means we get this:

..and that's exactly what we see on the output scope.

The rippling you see around sharp edges in a bandlimited signal is
called the [[WikiPedia:/Gibbs_phenomenon|Gibbs effect]]. It happens whenever you slice off part of the
frequency domain in the middle of nonzero energy.

The usual rule of thumb you'll hear is "the sharper the cutoff, the
stronger the rippling", which is approximately true, but we have to be
careful how we think about it.
For example... what would you expect our quite sharp anti-aliasing filter
to do if I run our signal through it a second time?

Aside from adding a few fractional cycles of delay, the answer is...
nothing at all.  The signal is already bandlimited. Bandlimiting it
again doesn't do anything.  A second pass can't remove frequencies
that we already removed.

And that's important.  People tend to think of the ripples as
a kind of [[WikiPedia:Sonic_artifact|artifact]] that's added by anti-aliasing and [[WikiPedia:Reconstruction_filter|anti-imaging]]
filters, implying that the ripples get worse each time the signal
passes through.  We can see that in this case that didn't happen. So
was it really the filter that added the ripples the first time
through?  No, not really. It's a subtle distinction, but Gibbs effect
ripples aren't added by filters, they're just part of what a
bandlimited signal ''is''.

Even if we synthetically construct what looks like a perfect digital
square wave,

it's still limited to the channel bandwidth.  Remember,
the stairstep representation is misleading.

What we really have here are instantaneous sample points,

and only one bandlimited signal fits those points.  All we did when we
drew our apparently perfect square wave was line up the sample points
just right so it appeared that there were no ripples if we played
[[WikiPedia:Interpolation|connect-the-dots]].

But the original bandlimited signal, complete with ripples, was
still there.

And that leads us to one more important point.  You've probably heard
that the timing precision of a digital signal is limited by its sample
rate; put another way,

that digital signals can't represent anything that falls between the
samples.. implying that [[WikiPedia:Dirac_delta_function|impulses]] or
[[WikiPedia:Synthesizer#ADSR_envelope|fast attacks]] have to align exactly
with a sample, or the timing gets mangled... or they just disappear.

At this point, we can easily see why that's wrong.

Again, our input signals are bandlimited. And digital signals are
samples, not stairsteps, not 'connect-the-dots'.  We most certainly
can, for example, put the rising edge of our bandlimited square wave
anywhere we want between samples.

It's represented perfectly and it's reconstructed perfectly.

==Epilogue==

[[Image:Moffey.jpg|360px|right]]

Just like in [[Videos/A_Digital_Media_Primer_For_Geeks|the previous episode]], we've covered a broad range of
topics, and yet barely scratched the surface of each one.  If anything, my
sins of omission are greater this time around... but this is a good
stopping point.

Or maybe, a good starting point.  Dig deeper.  Experiment.  I chose my
demos very carefully to be simple and give clear results. You can
reproduce every one of them on your own if you like.  But let's face
it, sometimes we learn the most about a spiffy toy by breaking it open
and studying all the pieces that fall out.  And that's OK, we're
engineers.  Play with the demo parameters, hack up the code, set up
alternate experiments.  The source code for everything, including the
little pushbutton demo application, is up at xiph.org.

In the course of experimentation, you're likely to run into something
that you didn't expect and can't explain.  Don't worry!  My earlier
snark aside, Wikipedia is fantastic for exactly this kind of casual
research. And, if you're really serious about understanding signals,
several universities have advanced materials online, such as the
[http://ocw.mit.edu/courses/electrical-engineering-and-computer-science/6-003-signals-and-systems-spring-2010/index.htm 6.003]
and
[http://ocw.mit.edu/courses/electrical-engineering-and-computer-science/6-007-electromagnetic-energy-from-motors-to-lasers-spring-2011 6.007]
Signals and Systems modules at MIT OpenCourseWare. And of
course, there's always the [http://webchat.freenode.net/?channels=xiph community here at Xiph.Org].

Digging deeper or not, I am out of coffee, so, until next time, happy
hacking!

==Credits==

Written by: Christopher (Monty) Montgomery and the Xiph.Org Community

Special thanks to:
*Heidi Baumgartner, for the second Tektronix oscilloscope
*Gregory Maxwell and Dr. Timothy Terriberry, for additional technical review

Intro, title and credits music:<br>
"Andy Warhol Is Gone", by Lousy Robot<br>
Used by permission of Lousy Robot.<br>
Original source track All Rights Reserved.<br>
[http://www.lousyrobot.com www.lousyrobot.com]

This Video Was Produced Entirely With Free and Open Source Software:<br>

*[http://www.gnu.org/ GNU]<br>
*[http://www.linux.org/ Linux]<br>
*[http://fedoraproject.org/ Fedora]<br>
*[http://cinelerra.org/ Cinelerra]<br>
*[http://www.gimp.org/ The Gimp]<br>
*[http://audacity.sourceforge.net/ Audacity]<br>
*[http://svn.xiph.org/trunk/postfish/README Postfish]<br>
*[http://gstreamer.freedesktop.org/ Gstreamer]<br>

All trademarks are the property of their respective owners. 

*''Complete video'' [http://creativecommons.org/licenses/by-sa/3.0/legalcode CC-BY-SA]<br>
*''Text transcript and Wiki edition'' [http://creativecommons.org/licenses/by-sa/3.0/legalcode CC-BY-SA]<br>

A Co-Production of Xiph.Org and Red Hat, Inc.<br>
(C) 2012-2013, Some Rights Reserved<br>

== Use The Source Luke ==

As stated in the Epilogue, everything that appears in the video demos is driven by open source software, which means the source is both available for inspection and freely usable by the community.  The Thinkpad that appears in the video was running Fedora 17 and Gnome Shell (Gnome 3).  The demonstration software does not require Fedora specifically, but it does require Gnu/Linux to run in its current form.

=== The Spectrum and Waveform Viewer ===

The realtime software spectrum analyzer application that appears in the video was a preexisting application that was dusted off and updated for use in the video.  The waveform viewer (effectively a simple software oscilloscope) was written from scratch making use of some of the internals from the spectrum analyzer application.  Both are available from Xiph.Org svn:

<center><div style="background-color:#DDDDFF;border-color:#CCCCDD;border-style:solid;width:80%;padding:0 1em 1em 1em;text-align:left;">
*Source for the Spectrum and Waveform applications is found at:
https://svn.xiph.org/trunk/spectrum/
*The source can be checked out of svn using the following command line:
svn co https://svn.xiph.org/trunk/spectrum
*Trac is a convenient way to browse the source without checking out a copy:
https://trac.xiph.org/browser/trunk/spectrum
</div></center>

Spectrum and Waveform both expect an input stream on the command line, either as raw data or as a WAV file.

=== GTK-Bounce ===

The touch-controlled application used in the video is named 'gtk-bounce' and was custom-written for the sole purpose of the in-video demonstrations.  It is so named because, for the most part, all it does is read the input from an audio device, and then immediately write the same data back out for playback.  It also forwards a copy of this data to up to two external monitoring applications, and in several demos, applies simple filters or generates simple waveforms. It includes several demos not included in the video.

<center><div style="background-color:#DDDDFF;border-color:#CCCCDD;border-style:solid;width:80%;padding:0 1em 1em 1em;text-align:left;">
*Source for gtk-bounce is found at:
https://svn.xiph.org/trunk/Xiph-episode-II/bounce/
*The source can be checked out of svn using the following command line:
svn co https://svn.xiph.org/trunk/Xiph-episode-II/bounce/
*Trac is a convenient way to browse the source without checking out a copy:
https://trac.xiph.org/browser/Xiph-episode-II/bounce/
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The application is somewhat hardwired for specific demo purposes, but most of the hardwired settings can be found at the top of each source file.  As found in SVN, the application expects an ALSA hardware audio device at hw:1, and if none if found, it will wait for one to appear.  Once a sound device is successfully initialized, it expects to find and open two pipes named pipe0 and pipe1 for output in the current directory.  In the video, the waveform and spectrum applications are started to take input from pipe0 and pipe1 respectively.  The output sent to the two pipes is identical, and in most demos matches the output data sent to the hardware device for conversion to analog.  The only exception is the tenth demo panel (which does not appear in the video) where gtk-bounce can be set to monitor the hardware inputs instead while the outputs are used to produce test waveforms.

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Assuming gtk-bounce, spectrum and waveform have been checked out and built, the configuration seen in the video can be started using the following commands:
* make the pipe fifos for the applications to communicate (only needs to be done once)
mkfifo pipe0; mkfifo pipe1
* start all three applications
waveform pipe0 & spectrum pipe1 & gtk-bounce &
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