Notes on DPS sliddeshow

Recording Using Digital Audio Tape Digital audio tape is a linear storage medium.  Each audio sample is recorded or striped on the tape one bit after the other, making it physically necessary to roll-back to an earlier place on the tape for music play-back.  Digital audio tape is primarily associated with VCR style Alesis’s  8-track ADAT and Tascam’s DA88 digital recorders and the 2-track DAT machines. Recording Using Internal RAM Internal RAM based recording is primarily associated with hardware keyboard samplers. The audio or sample data sits in active memory and is easily accessed, processed and manipulated.  However, recording time is limited to the total amount of internal RAM available in the device.  Because RAM chips are rather costly and limited in memory storage size, only a few minutes of stereo recording time is possible.  This makes internal RAM based recording useless for long real-time concert performances.

Recording Using a Hard-disk Drive. Recording using a hard-disk drive or SD card is currently the most effective and popular form of digital audio recording.  Hard-disk recording allows for instantaneous access to the audio data anywhere on the disk.  The amount of recording time is determined by the size of the drive or card.  Because audio data stored on a drive is bussed through the computer’s CPU, processing the audio data is fast and easy.  Audio files can be duplicated, preserving the original master recording. Hard-disk recording requires lots of unused disk space.  Since audio data is continuous, large amounts of  free or contiguous blocks of memory is necessary.  Therefore, using an external drive is critical when recording non-stop audio.  A dedicated drive, separate from the drive that contains the computer’s operating system and audio application, is highly recommended.  In this way, when reformatting or de-fragmenting the recording drive (a standard maintenance procedure), the computer’s operating system and audio application will not need to be reloaded.

Audio, in the form of analogue voltage, enters an anti-aliasing/low-pass filter with a cut-off frequency of 20kHz.  It then passes through an analogue to digital converter where the voltage is converted (at a given sampling rate and bit depth) into binary data.  The data then enters the computer’s CPU where it can be processed, saved to or loaded from an external memory storage device. From the computer, the data is converted back into an analogue voltage and again passed through a matching anti-aliasing/low-pass filter.  It is then sent out the analogue audio output to the audio sound system.

Bit Depth: The bit depth is the number of vertical bits used to represent the amplitude range of the audio data. The higher the bit depth: 8,16,24, the greater the overall dynamic range of the audio signal. The bit number represents the total amount of bits in a given linear bite. 

Sample rate: the sample rate is the frequency per second at which an incoming analouge voltage is converted into digital info or samples. The sample rate represents the horizontal time period of audio data. The higher the sampling rate, the greater the overall resolution of the audio signal resulting in a higher freuency response.

Nyquist theorem: if the sampling rate is at least 2.5 times the highest expected analogue input frequency, then the A/D and D/A converters will reproduce the audio signal with no distortion. 

1 bit= 2 outcomes2= 4 outcomes4= 16 outcomes8= 256 outcomes16= 65536 outcomes24= 16777216 outcomes

.aif= AUDIO INTERCHANGE FILE FORMAT, an uncompressed stereo/mono digital audio file format common on mac computers.

.sd2= SOUND DESIGNER 2, an uncompressed mono/stereo digital audio file format unit to didgidesign's audiomedia and pro tools systems.

.wav= an abbreviation for wave or waveform. an uncompressed digital audio file e format, common on PCs, was jointly developed by microsoft and IBM. used in both pc/mac.

MIDI CLOCK: sets the time-based to the speed at which a MIDI message is transmitted, 31.25kHz. uses Real Time.SAMPLE CLOCK: sets the time bass to the speed of the sample rateWORD CLOCK: the way in whcih a digital audio device sends a complete binary word.

digital audio= comprised of 0s and 1s, binary. Fixed DSP= the process is applied destructively, to a portion or other audio files. the audio is permently changed unless undo is selected immediately.  

Real Time DSP: The process is applied to an audio signal as the file is being played in real time. The audio is NOT permently changed.  

Normalize: To maximize the dynamic range of the audio waveform.  The process begins by searching for the highest dynamic or peak level.  This peak level is then increased to the full range of the bit depth.  The amplitude of the entire waveform is then increased by the same proportions. 

Fade In: To shape, according to an attack envelope, a portion or the entire waveform's amplitudeFadeout: To shape, according to a decay envelope, a portion ot the entire waveforms amplitudeChange Gain: to inscrese or decrease a portion or the entire waveform's amplitude by a set dB amount.Change Duration: to shorten or lengthen the duration of a waveform by a set time intervalChange Pitch: To change the fundamental frewuency of a waveform by a set interval. 

Reverse: to reverse a portion or the entire audio waveform, similar to playing an analogue backwardsInvert: to invert or flip a portion or the entire waveform up side down. this process is useful for fixing an out-of-phase audio file. 

Reverb: This group of plug-ins provides depth to the audio signal.  They are used to simulate various types of chambers, i. e., room, hall, stage, etc.  Controls include Room Shape, Room Size, Reverb Time, Reverb Pre-delay, Reverb Diffusion, High and Low Cut filter settings and Wet/Dry (processed to unprocessed) signal Mix. 

Delay: This group of plug-ins provides a range of time delays between the original signal and the delay signal.  The delay signal can have multiple generations known as “Feedback” creating an echo effect.  Controls include Delay Time with rhythm note values and Groove control, Feedback with Crossfeed control, High and Low Cut filters and Mix control. 

EQ:

EQ This group of plug-ins provides a variety of EQ systems.  Each system offers a unique way of making EQ adjustments to the audio signal.  There are several parametric EQs with graphic displays and multiple bands.  Controls include, Frequency, Amplitude and Cue controls (band width) with selectable EQ type presets.:

Dynamic: This group of plug-ins provides an assortment of dynamic processors such as Compressors, Expanders, Noise Gates and Envelopes.  Dynamic processing allows for control of the audio signal’s dynamic range and overall envelope shape. The audio signal can be compressed (as input gain increases, output gain decreases). Limited (as input gain increases, output gain remains fixed) and/or Gated (as input gain drops below a set value, the output signal is terminated).  Controls include Input and Output Level controls, Compression Ratio control, Threshold control, Attack and Release controls, and Knee control.  

Distortion: This group of plug-ins provides several distortion plug-ins such as Distortion, Overdrive and Bit-Crusher.  Controls include Drive, Tone, Output, Resolution, Down-Sampling, Clip Mode and Clip Level. 

Modulation This group of plug-ins provides several modulation effects such as Chorus, Flange, and Phase.  A modulation effect is created by oscillating the effect signal against the original signal, then mixing the result at the audio output. In the case of Phase Modulation, the phase angle of the waveforms’ amplitude is modulated by a sine wave LFO.  If the Phase modulation is stereo, a separate LFO for the left and right channel is provided. In the case of Flange modulation, the frequency component of the waveform is modulated by a sine LFO.  Color and Feedback levels are provided to increase harmonic content within the modulated signal. Modulation controls include: Mix, Intensity, Speed, Feedback, Color, Order and Stereo Phase.:

Waveform:A two-dimensional graphic representation of a mono or stereo sound wave.  The waveform’s vertical axis represents amplitude while the horizontal axis represents frequency and time. Selection:A shaded area created by dragging the mouse pointer/cursor along the length of the audio waveform. Region:A user-definable virtual amount of audio data with a variable start and end point.  Regions have no effect on the master audio file and are components of non-destructive editing. Playlist: A self-contained location where audio Regions are stored.  The Playlist may also serve as an audio sequencer for playing back Regions that have been placed in some sequential arrangement.  Generally, the Playlist displays the name of the region, its start and end times, its length, and its transition type.

Bouncing: An audio file that has been created by converting a Play List or a selected series of Regions into a completely new audio file. Destructive Editing:A type of editing process that permanently changes the audio material in some way. Nondestructive editing:A type of editing process that changes the audio material without permanently destroying the original audio file. Loop: A small section of an audio waveform that loops back to that waveform’s beginning.  A loop can be repeated for a designated number of times or indefinitely. Marker: A graphic placeholder that determines the start, end or loop point within a Region.  Markers can be inserted, moved, and deleted at any point along the waveform.

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Bit/sample/nyquist

DSP terms

DSP Terms 2