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  1. 1. Sound: Audio and MusicB.Sc. (Hons) Multimedia Computing Media Technologies
  2. 2. Agenda  History and overview of sound systems  Digitised Sound  Sound Formats  Sound for Multimedia  Software: Sound EditorsB.Sc. (Hons) Multimedia Computing Media Technologies
  3. 3. Edison Cylinder record
  4. 4. Earliest recorded sounds Thomas Edison’s  Au Clair de la voice (1890’s) Lune  Vocal Scale (Made by Scott, 1860)
  5. 5. Kymograph recording
  6. 6. How does it work?A diagonal movement provides either the LEFT or the RIGHT channel.A left-right movement provides the SUM of the L and R channels.An up-down movement provides the DIFFERENCE between the L and R channels.
  7. 7. Compact Disc
  8. 8. What’s in a CD? A. A polycarbonate disc layer has the data encoded by using ‘pits’. C. A reflective layer reflects the laser back. E. A lacquer layer is used to prevent oxidation G. Artwork is screen printed on the top of the disc. I. A laser beam reads the polycarbonate disc, is reflected back, and read by the player.
  9. 9. Encoding format on an audio CD (1) (Courtesy Wikipedia) The smallest entity in a CD is called a frame. A frame consists of 33 bytes and contains six complete 16-bit stereo samples (2 bytes × 2 channels × six samples: equals 24 bytes). The other nine bytes consist of eight Cross-Interleaved Reed-Solomon Coding (CIRC) error correction bytes and one subcode byte, used for control and display. Each byte is translated into a 14-bit word using Eight-to- Fourteen Modulation, which alternates with 3-bit merging words. In total there are 33 × (14 + 3) = 561 bits. A 27-bit unique synchronization word is added, so that the number of bits in a frame totals 588 (of which only 192 bits are music).
  10. 10. Encoding format on an audio CD (2) (Courtesy Wikipedia) These 588-bit frames are in turn grouped into sectors. Each sector contains 98 frames, totaling 98 × 24 = 2352 bytes of music. The CD is played at a speed of 75 sectors per second, which results in 176,400 bytes per second. Divided by 2 channels and 2 bytes per sample, this results in a sample rate of 44,100 samples per second.
  11. 11. Announced January 27th 2010…
  12. 12. Audio vs. Music  Computer sounds can be digital (e.g. .mp3) or synthesised (e.g. MIDI)  Digital sound is referred to as Audio  Synthesised sound is referred to as Music  Digital sounds are recordings of real sounds  Synthesized sounds are programmed reproductions of sounds based on algorithms and hardware tone generators.B.Sc. (Hons) Multimedia Computing Media Technologies
  13. 13. •Digitized Sound  Digital sound involves sampling, which means the encoding of data in the form of ones and zeros.  The computer converts an analogue data source to a digital data stream with an analog- to-digital converter.  These converters are devices that exist on the computers sound cards/modules, and are controlled by the software you use to process the sound, SoundForge on the PC, Sound Edit on the MAC, for example.B.Sc. (Hons) Multimedia Computing Media Technologies
  14. 14. What is the optimum sample rate for anaudio / video channel?The sample rate for the transmission of data through anoisy channel of restricted bandwidth was established byShannon and Weaver in 1948… Signal of amplitude 21 Plus noise of amplitude 3 Results in a noisy signal of amplitude 24.
  15. 15. How fast must we sample to accuratelyrepresent the signal?
  16. 16. Sample rate and resolution? There is no point in sampling  There is no point in sampling at a frequency higher than at a resolution greater than twice the bandwidth, that is, the maximum error – in this at 2 * W samples per case, the error is the noise, second. where W is the which is 3 units in 24 total bandwidth of the channel. units. Bandwidth could be thought  This gives us a resolution of as the frequency response (R)of 3 in 24 == 1 in 8 == 3 of the channel. bits.  So R=3=log2 Signal / Noise (log2 of 8 is 3)
  17. 17. Nearly there… Let’s call the sample rate R (Bits / Sec)… Since there are 2 * W samples per second, R = 2 * W * log2 (S+N)/N However, it’s more useful to measure power rather than amplitude (cables restrict power, not amplitude), and Power is proportional to the square of the amplitude, so… R = sqrt(2 * W * log2 (S+N)/N)
  18. 18. We’re there… R = sqrt(2 * W * log2 (S+N)/N)
  19. 19. An example… Let’s say we want to transmit human speech at a frequency range of 5Khz and a distortion (error) rate of 0.1% (i.e. 1 in 1000) R = 2 * 5,000 * log2 (1000+1)/1 = 50,000 bits/sec If we can tolerate an error of 4% (1000:4), then: R = 40,000 bits / sec. So, a small increase in error (distortion) rate has allowed a 20% reduction in the bandwidth of the channel (hence cost).
  20. 20. Synthesized Sound and MIDI  MIDI = Musical Instrument Digital Interface  Synthesized sound isnt digitally recorded; its a mathematical reproduction of a sound based on a description. Synthesizers use hardware and algorithms to generate sounds on-the-fly from a description of the desired sound.B.Sc. (Hons) Multimedia Computing Media Technologies
  21. 21. Sound Formats  Audio  .WAV (Developed by IBM and Microsoft): Uncompressed digital samples.  .AU  .AIFF (Audio Interchange File Format)  .MP3 (MPEG)  .SND (Mac)  Music  .MID (Musical Instrument Digital Interface)  and other proprietary formatsB.Sc. (Hons) Multimedia Computing Media Technologies
  22. 22. .mp3 (courtesy Wikipedia) The use in MP3 of a lossy compression algorithm is designed to greatly reduce the amount of data required to represent the audio recording and still sound like a faithful reproduction of the original uncompressed audio for most listeners. An MP3 file that is created using the setting of 128 kbits/s will result in a file that is about 1/11th the size of the CD file created from the original audio source. An MP3 file can also be constructed at higher or lower bit rates, with higher or lower resulting quality.
  23. 23. .mp3 (contd.) The compression works by reducing the accuracy of certain parts of sound that are deemed beyond the auditory resolution ability of most people. This method is commonly referred to as perceptual coding. It internally provides a representation of sound within a short-term time/frequency analysis window, by using psychoacoustic models to discard or reduce the precision of components less audible to human hearing, and recording the remaining information in an efficient manner (think about a handclap in a quiet room as oppose to in a noisy street –it’s masked by the noise).
  24. 24. Normal hearing thresholds…
  25. 25. Masked hearing thresholds…
  26. 26. Finally… Filters are used to split a signal into 32 bands, and a masking level for each band is computed. Signals that fall below the threshold can be discarded. Further compression is done, including resampling and lowering the bit rate of parts that don’t need full precision.
  27. 27. .mp3 file format
  28. 28. Sound for Multimedia Generate sounds for multimedia scenarios including:  Sound effects  Ambient sound for mood and effect  Virtual Juke Box  Voice-oversB.Sc. (Hons) Multimedia Computing Media Technologies
  29. 29. Software: Sound Editors  Sound Editors - edit waveforms of digitised sounds  Trim  Apply effects - e.g. reverb, echo, equalisation (EQ)  Insert silence  Fade-in, fade-out  Combine sounds  Adjust / Normalise volume (gain)B.Sc. (Hons) Multimedia Computing Media Technologies
  30. 30. Software: Sound Sequencers  Synthesized sounds played back via pre- recorded or live MIDI note event data - velocity, pitch, etc.  Sound cards may also have prerecorded digital samples stored on them.  These samples can then be played by sequencing software that will play a number of tracks simultaneously in an ensemble.  Software such as Cakewalk, Steinbergs Cubase, and Logic Pro.B.Sc. (Hons) Multimedia Computing Media Technologies
  31. 31. B.Sc. (Hons) Multimedia Computing Media Technologies
  32. 32. B.Sc. (Hons) Multimedia Computing Media Technologies
  33. 33. Hardware: Creative X-Fi•24-bit Analog-to-Digital conversion of analog inputs at 96kHz sample rate•24-bit Digital-to-Analog conversion of digital sources at 96kHz to analog 7.1speaker output•24-bit Digital-to-Analog conversion of stereo digital sources at 192kHz to stereooutput•ASIO (Audio Stream Input / Output) 2.0 support at 16-bit/44.1kHz, 16-bit/48kHz,24-bit/44.1kHz 24-bit/48kHz and 24-bit/96kHz with direct monitoring•16-bit to 24-bit recording sampling rates: 8, 11.025, 16, 22.05, 24, 32, 44.1, 48and 96kHz•Digital SPDIF interface support with 24bit/96kHz quality format•Signal-to-Noise ratio: 109dB•Total Harmonic Distortion + Noise at 1kHz = 0.004%
  34. 34. X-Fi board
  35. 35. X-Fi control panel
  36. 36. X-Fi software control panel
  37. 37. Please read… Chapman & Chapman Chapter 8 (in your bundle if you have it)