digital audio the nuts and bolts n.
Skip this Video
Loading SlideShow in 5 Seconds..
Digital Audio — The Nuts and Bolts PowerPoint Presentation
Download Presentation
Digital Audio — The Nuts and Bolts

Digital Audio — The Nuts and Bolts

0 Views Download Presentation
Download Presentation

Digital Audio — The Nuts and Bolts

- - - - - - - - - - - - - - - - - - - - - - - - - - - E N D - - - - - - - - - - - - - - - - - - - - - - - - - - -
Presentation Transcript

  1. Digital Audio —The Nuts and Bolts A digital audio overview ranging from bit rate, sample rate, and compression types to room acoustics, microphones, and digital effects

  2. Sound Waves/Analog Audio • Sound waves are continuous • Infinite number of amplitude points can be identified between any two points in time

  3. Digital Audio • Computers don’t deal with continuous concepts (infinity) • Digital technology converts analog audio to computer values

  4. Digital Conversion • Digitizing a continuous wave = sampling • Amplitude measurements of a sound signal are regularly sampled

  5. ADC and DAC • ADC – Analog to Digital Converter Converts analog signal to digital samples • DAC – Digital to Analog Converter Converts digital samples to analog signal

  6. Characteristics ofDigital Audio • Sampling Rate • How often signal is sampled • Number of samples per second • Bit Depth • Size of number used to store samples • larger number gives more degrees of value

  7. Sampling Rate • Harry Nyquist (Bell Labs – 1925) • Nyquist Theorem: To represent digitally a signal containing frequency components up to X Hz, it is necessary to use a sampling rate of at least 2X. • Humans hear to 20 kHz, requiring sample rate of at least 40k

  8. Aliasing • In movies, car wheels appear to move backwards if between ½ and 1 revolution per frame • In sound, this is not acceptable • Filters are used to remove any frequencies above Nyquist frequency

  9. Undersampling

  10. Undersampling = Aliases

  11. Critical Sampling

  12. Lowpass Filter • Reduces or eliminates higher frequencies • Used to remove any frequencies above Nyquist frequency

  13. Bit Depth (Quantization) • Amplitude values are stored as binary numbers • Accuracy depends on how many bits are available to represent these values • For CD Audio we use 16 bits

  14. Quantization • No matter how many bits are used, there is always a margin of error • Low-level signals do not use all available bits, so signal-to-error ratio is greater

  15. Quantization • Quantization error creates a kind of distortion • Dither adds low-level noise to audio signal before sampling • Dither turns distortion (bad) into noise (less bad) – still less noise than analog

  16. Digital Recording Process • Dither – Low-level noise added (prior to sampling) to reduce quantization error distortion

  17. Digital Recording Process • Lowpass Filter – Removes frequencies above Nyquist Frequency; cutoff starts a few thousand hertz lower

  18. Digital Recording Process • Sample and Hold – Analog voltages are measured and held long enough to be read by ADC

  19. Digital Recording Process • Analog-to-Digital Converter – Converts analog voltages into binary numbers

  20. Digital Recording Process • Multiplexer – Combines the parallel data streams (stereo) into a single serial bit stream

  21. Digital Recording Process • Error Correction – Variety of measures to eliminate, reduce, or compensate for errors

  22. Digital Recording Process • Encoding – Encoded for playback

  23. Digital Recording Process • Storage

  24. Digital Playback Process • Buffer – To ensure that samples are processed at a constant rate

  25. Digital Playback Process • Error Correction – Attempt to eliminate, reduce, or conceal data errors

  26. Digital Playback Process • Demultiplexer – Splits the serial bitstream into parallel data streams (stereo)

  27. Digital Playback Process • DAC – Digital-to-Analog converter translates binary numbers to voltage values

  28. Digital Playback Process • Sample and Hold – Reads the value from the DAC and holds it until the DAC’s next stable state

  29. Digital Playback Process • Lowpass Filter – Smooths the output from the sample and hold circuit

  30. Digital Playback Process • Audio – The finished product

  31. Room Acoustics • Characteristic room sound is determined by the relationship between direct and reflected sound • Virtually all sound reaching listeners is a combination of direct & reflected • At greater distances, most sound is reflected sound

  32. Room Acoustics • Direct Sound • Directly from the source to the listener • Direct sound arrives before reflected sound; even if reflected sound is louder, we hear direct sound first and determine direction of the source

  33. Room Acoustics • Early Reflections • First-order reflections that reach the listener after reflecting once from the floor, ceiling, or walls • If arriving in the first 35ms after the direct sound, reinforces with clarity & intelligibility • “Intimate” halls have first-order reflections of less than 20ms

  34. Room Acoustics • Diffuse Reverberations • Second- (and higher) order reflections • Reverberation time is the time required for the SPL to drop 60dB • Larger room is likely to have longer reverberation time than a smaller room • Reverberation time is frequency dependent; lower frequencies reverberate longer

  35. Types of Reflections • Specular • Reflections off smooth and regular surfaces • reflection in one direction • Diffuse • Reflections off irregular surfaces • Reflections scattered in many directions • Contribute to sound of older concert halls

  36. Absorption

  37. Small Room • Space has potential to act as closed tube, producing standing wave • Result is amplification of certain frequencies based on room’s dimensions • Not a factor in large rooms because air temperature varies more

  38. Microphones • Receptor type • Diaphragm acts as receptor • Diaphragm vibrates • Transducer type • Transducer converts vibrations to electricity • Directionality • Determines strength of signal produced by sounds arriving from different directions

  39. Receptor Types • Pressure • Diaphragm responds to sound pressure changes on only one side of diaphragm • Pressure Gradient • Diaphragm responds to sound pressure changes from the front or rear • Signal is determined by difference (gradient) of pressures from either side

  40. Transducer Types • Dynamic (Electrodynamic, Electromagnetic, Ribbon, Moving Coil) • Principle of magnetic induction – wire moves within a magnetic field, producing a current • Inexpensive and sturdy • Condenser (Capacitor) • Two oppositely-charged metal plates • Current moves from one to the other • Sharper transients • Expensive

  41. Directionality • Determines the strength of signal produced by sounds arriving from different directions • Directionality varies with frequency • Specs often include polar plot with patterns for different frequencies

  42. Omnidirectional • Responds equally to sound from all directions • Pressure mics are omnidirectional

  43. Bidirectional • Figure-eight response • Responds equally to sounds from front & back; none from sides • Pressure gradient mics are bidirectional

  44. First-Order Cardioid • Most common directional microphones • Cardioid refers to heart-shaped pattern • Directional patterns are obtained by combining pressure and pressure gradient elements in varying proportions

  45. Cardioid Variations 50% Pressure/50% Pres. Gradient 75% Pressure/25% Pres. Gradient 37% Pressure/63% Pres. Gradient 25% Pressure/75% Pres. Gradient

  46. Effects • All music that is recorded or amplified relies on effects to enhance the sound. • Effects are necessary to make electronic audio signals sound like natural sound.

  47. Effects = Filters • Effects are created by filter combinations • Filtering involves combining original signal with delayed version • Higher internal processing bit rate means more accurate arithmetic

  48. Simple Delay • Signal combined with delayed version of itself.

  49. Multitap Delay • Series of Simple Delays; output is combines with a succession of delays.

  50. Feedback Delay • Combines delayed output with input, then sends through delay again.