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Exploring Audio Compression and Streaming Techniques

This text delves into various aspects of audio file compression, streaming, and the intricacies of loudness and signal-to-noise ratios in digital audio. Dr. Paul Vickers discusses key concepts such as the power of sound, sampling errors, quantization, and the significance of dynamic range in audio resolution. He examines coding methods including PCM, DPCM, and ADPCM, highlighting lossless versus lossy compression techniques. Also noted are the challenges of compressing audio effectively due to its random noise components, as well as innovative methods like sub-band and speech compression.

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Exploring Audio Compression and Streaming Techniques

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  1. Compression & Streaming Serving, shrinking, and otherwise messing about with perfectly good audio files Dr Paul Vickers

  2. Loudness and power • Loudness related to force with which a sound presses on your eardrum • The more power, the louder the sound • Power is proportional to the square of a sound’s intensity (amplitude, or voltage) Dr Paul Vickers

  3. Sampling error and noise • CD audio uses 44.1 KHz at 16 bit resolution • Sampled voltages quantised to –32768…32767 • Quantisation introduces error (through rounding) • Largest error is 0.5 which is 2-16 times as loud as the loudest sample value • Power related to square of amplitude so error has power 2-32 as loud as loudest signal • Ratio of signal to error (noise) is 232:1 • Or 96.3 dB (10 log10(232)) • = SNR of 96 dB Dr Paul Vickers

  4. Signal to noise ratio • So, CD audio has SNR of 96 dB • 8-bit sampling has SNR of 48 dB • Therefore, 1 bit of resolution adds approx. 6 dB to the dynamic range • Threshold of pain is 120 dB so we need a 20-bit resolution to capture the dynamic range of human auditory system • Loud samples are rare, so noise is more noticeable than the theory would suggest Dr Paul Vickers

  5. Coding • A standard .WAV file (no such thing) stores samples as 16-bit values. • These values are codes representing the voltages (amplitudes) of the signal • System called pulse code modulation (contrast with pulse amplitude modulation and pulse width modulation) • WAV format actually supports nearly 100 different coding systems Dr Paul Vickers

  6. Compression • Lossless compression (e.g. LZW) does not work well on audio as there are very few repeating patterns • Sampled audio tends to have random noise in the least significant bits making very few bytes identical. • Winzip hardly compresses audio files at all • Try girl2.wav and 528 Hz.wav. Why does the second file compress 2.33:1? Dr Paul Vickers

  7. Other techniques • Need some different compression techniques • Popular ones are: • Differential PCM (DPCM) • Adaptive DPCM (ADPCM) • A-Law • µ-Law • Logarithmic & non-linear codings • Perceptual codings Dr Paul Vickers

  8. Differential PCM • Consider the differences in value between individual samples at rates of, say, 44.1 KHz • Usually fairly small • Small differences need fewer bits than the samples themselves • So, DCPM stores sample differences, hence the name • Leads to some inaccuracy and requires look ahead to balance things out Dr Paul Vickers

  9. DCPM example • To reduce 8-bit sample values to 4-bit differences • Consider three samples of 17, 28, 30 • Differences: 11, 2 • 4-bit system only allows values -8…+7 (1000…0111) • Thus 11 overflows, therefore clipped at 7 • But decompressing would then give 17, 24, 26 • But if we look at diff. between decompressed sample and next actual: • 17-28 = 11 -> 7. 17 + 7 = 24. Diff. 24-30 = 6 • Give 7, 6 which, when decompressed gives 17, 24, 30 Dr Paul Vickers

  10. Predictor based compression • Try to predict next sample on basis of previous samples • If correct, no need to store sample as decompressor uses same rules and so can work it out too • If prediction correct, output 1 else output 0 followed by actual sample Dr Paul Vickers

  11. Adaptive DPCM • ADPCM uses prediction • Outputs predicted differences. If accurate then diff between actual and predicted samples has lower variance than actual samples and thus take fewer bits • Uses 4-bit codes representing predicted diff. between two 16-bit samples Dr Paul Vickers

  12. Sub-band coding • Low frequencies have fewer cycles per second and thus lots of small differences • High frequencies have larger differences • Dividing signal into frequency bands allows low frequencies to be coded with fewer bits than high frequencies • Bands to which ear is less sensitive can be less accurately stored Dr Paul Vickers

  13. Speech compression • Musical sound has little silence • Speech has many pauses and silences • These can be replaced by duration codes • Can reduce a signal by 50% by doing this Dr Paul Vickers

  14. Checkpointing • Predictive techniques need knowledge of what has gone before • If a stream (e.g. love radio feed) is opened in the middle, this state information is unavailable • Therefore, insert checkpoints that contain • Uncompressed samples, or • Compressor state vector • Checkpoints allow decompressor to reset itself Dr Paul Vickers

  15. Non-linear coding • High sample rate gives wide dynamic range • Reducing from 16 bits to 8 bits halves storage requirements, but reduces dynamic range by 63,000 times (96 dB down to 48 dB) • Standard PCM is linear • Sample value 50 is twice the amplitude of 25 • In 8-bit system, sounds less than 1/256th of loudest possible signal disappears Dr Paul Vickers

  16. Non-linear coding • Ear is quite insensitive to small changes in loud sounds but very sensitive to same small change in quieter sounds • Linear coding ideal of computational manipulation but wasteful • Non-linear coding uses a logarithmic scale • Value of 1 may be much less than 1/50th of intensity represented by value of 50 • More bits for quiet sounds and fewer bits for very loud sounds Dr Paul Vickers

  17. -Law & A-Law • -Law and A-Law uses logarithmic compression to convert linear-coded PCM samples into 8-bit codes • Provide greater accuracy for the small (quiet) samples that form bulk of an audio signal • Human auditory system has (approx) logarithmic response so these techniques give highest accuracy where most audible • Dynamic range is 14 bits & 13 bits respec. (84 dB and 78 dB) Dr Paul Vickers

  18. Perceptual coding • DPCM, ADPCM, -Law & A-Law do not give high-enough compression for demanding multimedia and web applications • Using psychoacoustic models of our auditory system we can take information out of the audio signal without changing its perceptual characteristics (well, sort of) • Linear PCM captures sound as it is • Perceptual coding captures audio as it sounds Dr Paul Vickers

  19. Perceptual coding • PC uses knowledge of the masking properties of the human auditory system and our sensitivity to different frequency bands • PC introduces significant noise into the signal… • … but in such a way as we don’t hear it. • MP3, ATRAC (mini disc), DCC use perceptual coding techniques Dr Paul Vickers

  20. Masking • Part of an audio signal can be inaudible • A loud sound can mask a simultaneous quiet sound • A quiet sound immediately following a very loud sound may also be inaudible • E.g. you have to turn up the radio when your car goes faster • E.g. A handclap (normally loud) heard straight after a gun shot would sound quiet • PC assigns fewer bits to masked signals Dr Paul Vickers

  21. MPEG audio • MPEG audio layer 1, 2, & 3 • Most commonly use layer 3, hence MP3 • A standard for coding an audio stream into a bit stream at various bit rates • The higher the bit rate, the more data • At a bit rate of 96 kpbs achieve bandwidth of about 15 KHz and compression of 16:1 • At 128 kpbs, get closer to 20 KHz and compression of about 12:1 Dr Paul Vickers

  22. ATRAC • Mini disc uses adaptive transform acoustic coding • Compression of 5:1 • Like MP3 uses perceptual coding and sub-band compression • ATRAC uses three sub-bands, MP3 uses 32 Dr Paul Vickers

  23. Streaming • Streaming is the process of sending an audio file as a continuous stream that can be played back the moment the stream starts • Avoids having to download the file first • suitable for live situations, e.g. web casts, internet radio, etc. • Need to know about network capabilities of client • e.g. no point sending 128 kbps MP3 audio to a 56 k modem client Dr Paul Vickers

  24. Streaming • Smooth signal heard where transmitter sends data at least as fast as client can decode it • low bandwidth connections and • network congestion lead to low stream rate = either poorer quality audio, or glitches and pauses • Popular formats are Real audio, MS ASF, Apple Quicktime Dr Paul Vickers

  25. Creating streamed content • Very simple • Connect a live feed to a streaming-enable media producer • Use tools such as Windows Media Encoder or Real’s Helix Producer to turn audio files into streamable files. Even Sound Forge can save as .ASF and .RM • Select required bit rate/bandwidth • Some services provide multiple bit rates Dr Paul Vickers

  26. Example http://computing.unn.ac.uk/staff/cgpv1/music!.htm Dr Paul Vickers

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