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SPEAKER 1: In this
lab, your task is going

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to be to write a program in C that
changes the volume of an audio file.

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Specifically, in this
lab, we're going to be

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working with WAV files, a standard
file format for representing

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audio information.

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What is the structure of a WAV file?

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Well, like any file, these
files consist of bytes.

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And in particular, in a WAV file,
the first 44 bytes of that file

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are the header for that WAV file that
contain information about that file

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that your computer, or
programs on your computer,

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might need to know in order to read
and understand and interpret that file.

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After the 44-byte header, are
two byte samples of audio.

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Many, many of these
2-byte samples of audio

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are repeated back to back to back,
where each of these 2-byte samples

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represents the value of some audio
waveform at a particular point in time.

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So ultimately, your WAV file is going to
consist of one 44-byte header followed

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by many of those 2-byte samples.

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And the interesting thing
about those 2-byte samples

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is that each one is
really just a number.

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And if you want to change
the volume of a sample,

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you're going to multiply
that sample by a number.

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If I have an audio sample, and
I want to double the volume,

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for example, I would take each of the
samples of audio in the audio file

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that I'm trying to modify and just
multiply each of those sample values

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by 2.

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Likewise, if I'm trying
to cut the volume in half

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for a particular audio file, then
I'll take each of the 2-byte samples,

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and I'll multiply each by 0.5, or 1/2,
in order to cut the volume in half

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as well.

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Ultimately, the way your program is
going to work is we're going to run

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our program as ./volume followed
by three command line arguments.

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The first command line
argument is an input WAV file,

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some file that already exists
representing the audio file that we're

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here going to try to modify.

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The next command line argument
is the name of some output file,

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some new audio file that
our program is going

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to generate that is going
to have the modified volume.

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And finally, the last command
line argument to this program

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is going to be a floating
point number representing

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the factor by which to change the
volume of the original audio file.

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In this case, for example,
we're using a factor of 2.0

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to mean we want to double
the volume of the audio file.

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But if instead we had tried to use
0.5, for example, as the factor,

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then we would be cutting the
volume of the audio file in half.

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So here's what you're going
to need to do in this lab.

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In order to take the input
file and then generate

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a new output file that's the same
audio but just with the volume

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changed, you're going to want to start
with the header of that WAV file,

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since the header, those
first 44 bytes of the file

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are the first thing you'll
find inside the input WAV file.

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So you'll initially want to read
the header from the input file.

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And then you'll want to write that
exact same header to the output file.

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After the header, recall, comes
each of those 2-byte samples,

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many 2 byte samples,
back-to-back in the file,

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representing the audio
waveform of the file itself.

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So you'll want to repeat, for
each of those 2-byte samples,

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you'll want to first read the
sample from the input file,

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likely into some sort of variable.

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And then, you'll want to multiply that
sample value by the volume factor.

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If you're trying to double
the volume of the sample,

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then you'll double that volume.

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If you're trying to triple
it, you'll multiply it by 3.

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If you're trying to
cut the volume in half,

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you'll multiply by 0.5, for example.

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And after you've done
that multiplication,

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you'll then write that new
sample to the output file.

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So that the updated sample,
with the modified volume,

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ends up being written to the output
file for each of the samples of audio

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in that entire file.

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As you go about working on this lab, a
couple of techniques will be helpful.

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One is taking advantage of
the various different types

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that C gives you for dealing
with data of different sizes.

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uint8_t, for example, is a type
that stores unsigned integers that

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are 8 bits, or 1-byte large.

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And this is useful anytime you want to
represent just a generic byte of data.

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For example, if you're trying to read
in a header that has 44 bytes of data,

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you might use an array of 44 of
these unsigned integers of 8-bit size

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as uint8_ts, inside of an
array, to store your header.

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Meanwhile, int16_t is a type
that stores signed integers,

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meaning they could be positive or 0 or
negative, of 16-bits, or 2-bytes, each.

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And ultimately, this is a great choice
of type for representing your samples.

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Because we know that each
sample is 2 bytes large.

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And int16_t is the perfect size for
representing this kind of information.

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Next you're also going to want to
deal with reading and writing files.

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And for that, these two functions will
likely prove helpful, fread and fwrite.

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fread will read a certain
number of bytes from some file

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into memory inside of your computer.

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And likewise, fwrite can write
data from your computer's memory

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to a particular file.

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And you'll likely want to look
to the documentation for each

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of these functions, fread and fwrite to
get an understanding for what arguments

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they take in what order and how you
might use these functions to read

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from the input file and then write
your updated data to your output file.

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Now let's take a look
at the distribution code

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that we give to you as part of this lab.

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You'll notice that one
of the first things we

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give you is a constant integer
called HEADER_SIZE, equal to 44.

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You can use this
variable anytime you need

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to reference the number of bytes
inside of the header of a WAV file

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as this constant integer
called HEADER_SIZE.

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Inside of the main function, we've
already done some work for you.

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First, checking the number
of command line arguments

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to ensure that when the user
is running this program,

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they're providing an input
file, an output file,

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as well as some factor by which to
change the volume of the input file.

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We then open the input file,
checking to make sure that the input

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file was able to be opened correctly.

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We then open the output file using
the W mode, W for writing that file,

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in order to make sure that we
can open the file that we're

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going to be writing to correctly.

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And then we compute the
factor, converting that factor

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to a floating point number like
2.0 for doubling the volume or 0.5

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for cutting the volume in half.

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Here, then, are your to dos.

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The first thing you want to do is to
copy the header from the input file

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to the output file.

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Recall that you know that header
is always going to be 44 bytes.

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So you'll likely want to use fread
to read 44 bytes from your file

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and then fwrite to write 44
new bytes to the output file.

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Then you'll want to go
through each of the samples,

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likely using some kind of
loop to loop through the input

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file until you get to
the end of the file.

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And for each of those 2-byte samples, to
read it into memory, update the volume,

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and then write that updated
sample to the output file as well.

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You'll notice that we provide you
with a sample input file, input.wav,

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which you can listen to
as just a sample of audio.

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And you can run your volume program on
this file, passing in a factor of 2.0

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to double the volume,
or some other factor

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to change the volume
by a different amount.

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And then you should be able to
listen to that new output file

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and detect that it does, in
fact, have the same audio, just

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with a different volume.

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After you've done all
of those steps, you

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should then have a program
that you can take any WAV file

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and change the volume
by a particular factor.

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My name is Brian.

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And this was volume.

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