Audio Recording

A typical computer system works with digital audio in the following way. First, to record digital audio, you need a card with an Analog to Digital Converter (ADC) circuitry on it. This ADC is attached to the Line In (and Mic In) jack of your audio card, and converts the incoming analog audio to a digital signal that computer software can store on your hard drive, visually display on the computer's monitor, mathematically manipulate in order to add effects or process the sound, etc. (When I say "incoming analog audio", I'm referring to whatever you're pumping into the Line In or Mic In of your sound card, for example, the output from a mixing console, or the audio output of an electronic instrument, or the sound of some acoustic instrument or voice being feed through a microphone plugged into the sound card's Mic In, etc). While the incoming analog audio is being recorded, the ADC is creating many, many digital values in its conversion to a digital audio representation of what is being recorded. Think of it as analogous to a cassette recorder. While you're recording some analog audio to a cassette tape, the tape is constantly passing over the record head. So the longer the passage of music you record, the more cassette tape you use to record that analog audio signal onto the tape. So too with the conversion to digital audio. The longer the passage of music you record (ie, digitize), the more digital values are created, and these values must be stored for later playback.

Where are these digital values stored? Well, as your sound card creates each value, that data is passed to the card's software driver which then passes that data to the software program managing the recording process. Such software might be CakeWalk recording a digital audio track, or SAW, or Samplitude, or Windows Sound Recorder, or any other program capable of initiating and managing the recording of digital audio. Whereas the program may temporarily accumulate those digital audio values in the computer's RAM, those values will eventually have to be stored upon some fixed medium for permanence. That medium is your computer's hard drive. (For this reason, sometimes people refer to the process of recording digital audio to a hard drive as "Hard Disk Recording". Henceforth, I will abbreviate "hard drive" as HD). Usually, how this works is that the program accumulates a "block" of data in RAM (while the sound card is digitizing the incoming analog audio), for example 4,000 data values, and then writes these 4,000 values into a file on your HD. (It's a lot more efficient to write 4,000 values at once to your hard drive, than it is to write those 4,000 values one after the other separately). All of these 4,000 values go into one file. Usually, the format for how the data is arranged within this file follows the WAVE file format. (But there are other formats that may also be used by programs to store digital audio values. For example, AIFF is often used on the Macintosh. AU format is used on Sun computers. MP3 is a popular format nowadays because it compresses the data's size. Etc. Any of these formats could be used on any computer, but WAVE is considered the standard on a Windows-based PC). Now, if the recording process is still going on, the software program will collect 4,000 more values in RAM, and then write them out to the same WAVE file on the HD (without erasing the previously stored 4,000 values -- ie, the values accumulate within the WAVE file, so that it now has 8,000 values in it). This process continues until the musician halts the recording process. ( ie, If you let recording continue long enough, it will eventually fill up your HD with digital audio values in one, big WAVE file). At that point, the WAVE file is complete, and contains all of the digital audio values representing the analog audio recorded.

So, digital audio recorded by most PC programs is predominantly stored in a WAVE file on your HD, and that file is created while the digital audio is being created/recorded.

Audio Playback

Some programs can optionally perform some mathematical manipulation of the digital audio values immediately before the data is passed to the card's driver (ie, during playback). Such manipulations may be to add effects such as reverb, chorus, delay, etc. Programs that do such realtime processing are SAW and Samplitude.

Some programs also can process the digital audio to do some valuable things such as "time-stretching" and "pitch shift" (although these are often too computationally complex for today's computer to do during playback. So, this processing usually has to be applied before playback, and may be "destructive" in that it permanently alters the digital audio data). When you play a one-shot (ie, non-looped) waveform, it lasts only so long (ie, in terms of time). For example, maybe you've got digital audio tracks of a piece of music whose duration is 2 minutes. Sometimes, people need to adjust the length of time over which the waveform plays. For example, maybe the producer of a movie says "I want this piece of music to last exactly 2 minutes and 3 seconds in order to fit with this filmed scene I have which is this long". Well, if you had a MIDI track, you'd just slow the tempo a little in order to make the music last that extra 3 seconds. The music would sound exactly the same, but at a slightly slower tempo. OK, so how do you "slow down" the digital audio tracks? Well, you could reduce the playback rate a little. But, if you've ever used a sampler, you'll notice that when you play a waveform at a rate different than it was recorded, this changes the character of the waveform itself. You don't just hear a different tempo, you hear different pitch, vibrato, tremulo, tone, etc. So, time-stretching was devised to take a waveform, analyze it, and change its length without (hopefully) changing its characteristics. The net result is that you get the same effect that you had by changing the tempo of the MIDI track; ie, merely a change in tempo/duration rather than a change in the characteristics of the waveform. (Nevertheless, there is always some potential for a change in timbre when time-stretching algorithms are applied to a waveform).

Time-stretching: When you play a one-shot (ie, non-looped) waveform, it lasts only so long (ie, in terms of time). For example, maybe you've got digital audio tracks of a piece of music whose duration is 2 minutes. Sometimes, people need to adjust the length of time over which the waveform plays. For example, maybe the producer of a movie says "I want this piece of music to last exactly 2 minutes and 3 seconds in order to fit with this filmed scene I have which is this long". Well, if you had a MIDI track, you'd just slow the tempo a little in order to make the music last that extra 3 seconds. The music would sound exactly the same, but at a slightly slower tempo. OK, so how do you "slow down" the digital audio tracks? Well, you could reduce the playback rate a little. But, if you've ever used a sampler, you'll notice that when you play a waveform at a rate different than it was recorded, this changes the character of the waveform itself. You don't just hear a different tempo, you hear different pitch, vibrato, tremulo, tone, etc. So, time-stretching was devised to take a waveform, analyze it, and change its length without (hopefully) changing its characteristics. The net result is that you get the same effect that you had by changing the tempo of the MIDI track; ie, merely a change in tempo/duration rather than a change in the characteristics of the waveform. (Nevertheless, there is always some potential for a change in timbre when time-stretching algorithms are applied to a waveform).

Pitch shift: This changes the note's pitch without altering the playback rate (which would alter other characteristics of the waveform). So, if you sampled the middle C note of a piano at 44KHz, but when you play it back at 44KHz, you really want to hear a D note, you could apply pitch shift to it to create a waveform whose "root" (ie, the pitch you get when you playback at the same sample rate as when recorded) is D. If you take a musical performance with "rhythms" in it, and apply pitch shift, you should get a different pitch, but retain the same rhythms (ie, tempo).

Virtual Tracks