Audio Coding and

Standards
Audio Coding and Standards
• Models, Techniques & Requirements of Sound Coding
• Entropy Coding: Run length Coding & Huffman coding
• Differential Coding – DPCM & ADPCM
• LPC and Parametric Coding
• Sound Masking Effect and Sub-band Coding
• ITU G.72x Speech/Audio Standards
• ISO MPEG-1/2/4 Audio Standards
• MIDI and Structured Audio
• Common Audio File Formats
PCM Audio Data Rate and Data Size

Conclusion  Need better coding for compressing sound data

Models & Techniques of Sound
Compression
Requirements for Compression
Algorithms
• Lossless compression: y(n) = x(n)
• Decoded audio is mathematically equivalent to the original one
• Drawback : achieves only a small or modest level of compression

• Lossy compression: y(n) ≠ x(n)

• Decoded audio is worse than the original one -> Distortion
• Advantage: achieves very high degree of compression
• Objective: maximize the degree of compression in certain quality

• General compression requirements:

• Ensure a good quality of decoded/uncompressed audio
• Achieve high compression ratios
• Minimize the complexity of the encoding and decoding process
• Support multiple channels
• Support various data rates
• Give small delay in processing
Entropy Coding
• Entropy encoding (lossless): Ignores semantics of input data and
compresses media streams x(n) by regarding them as sequences of
digits or symbols
• Examples: run-length encoding, Huffman encoding , ...
Entropy Coding
• Run-length encoding:
• A compression technique that replaces consecutive occurrences of a symbol with the symbol
followed by the number of times it is repeated
• a a a a a => ax5
• 000000000000000000001111111 => 0x20 1x7
• Most useful where symbols appear in long runs: e.g., for images that have areas where the
pixels all have the same value, fax and cartoons for examples.
Entropy Coding
• Huffman encoding:
• A popular compression technique that assigns variable length binary codes to symbols, so that the most
frequently occurring symbols have the shortest codes
• Huffman coding is particularly effective where the data are dominated by a
small number of symbols, e.g.
• {x(n)} = hfeeeegheeegdeeehehcfbeeeeeqghf…
• Suppose to encode a source of N =8 symbols: X(n)->{a,b,c,d,e,f,g,h}
• The probabilities of these symbols are: P(a) = 0.01, P(b)=0.02, P(c)=0.05,
P(d)=0.09, P(e)=0.18, P(f)=0.2, P(g)=0.2, P(h)=0.25
• If assigning 3 bits per symbol (000~111), the average length of
symbols is: L=3bits
• The theoretical lowest average length – Entropy

• If we use Huffman encoding, the average length = 2.63 bits/symbol

Huffman Coding
• The Huffman code assignment procedure is based on a binary tree
structure. This tree is developed by a sequence of pairing operations
in which the two least probable symbols are joined at a node to form
two branches of a tree. More precisely:
• 1. The list of probabilities of the source symbols are associated with the
leaves of a binary tree.
• 2. Take the two smallest probabilities in the list and generate an intermediate
node as their parent and label the branch from parent to one of the child
nodes 1 and the branch from parent to the other child 0.
• 3. Replace the probabilities and associated nodes in the list by the single new
intermediate node with the sum of the two probabilities. If the list contains
only one element, quit. Otherwise, go to step 2.
Huffman Coding
Huffman Coding
• The new average length of the source:

• The efficiency of this code is:

• How do we estimate the P(i) ? Relative frequency of the symbols
• How to decode the bit stream ? Share the same Huffman table
• How to decode the variable length codes ? Prefix (free) codes have the property that no
codeword can be the prefix (i.e., an initial segment) of any other codeword. Huffman codes are
prefix codes !
• – 00000100100110 => ? beef
• Does the best possible codes guarantee to always reduce the size of sources? No. Worst case
exists. Huffman coding is better averagely.
• Huffman coding is particularly effective where the data are dominated by a small number of
symbols
Differential Coding – DPCM &
ADPCM
• Based on the fact that neighboring samples … x(n-1), x(n), x(n+1), … in a
discrete audio sequence changing slowly in many cases
• A differential PCM coder (DPCM) quantizes and encodes the difference

• Advantage of using difference d(n) instead of the actual value x(n)

• Reduce the number of bits to represent a sample
• General DPCM: d(n) = x(n) – a1x(n-1) - a2x(n-2) -…- akx(n-k)
• a1, a2, …ak are fixed
• Adaptive DPCM: a1, a2, …ak are dynamically changed with signal
LPC and Parametric Coding

• LPC (Linear Predictive Coding)

• Based on the human utterance organ model
• s(n) = a1s(n-1) + a2s(n-2) +…+ aks(n-k) + e(n)
• Estimate a1, a2, …ak and e(n) for each piece (frame) of speech
• Encode and transmit/store a1, a2, …ak and type of e(n)
• Decoder reproduce speech using a1, a2, …ak and e(n)
• - very low bit rate but relatively low speech quality
• Parametric coding:
• Only coding parameters of sound generation model
• LPC is an example where parameters are a1, a2, …ak , e(n)
• Music instrument parameters: pitch, loudness, timbre, …
 Sub-band Coding
• Human auditory system has limitations
• Frequency range: 20 Hz to 20 kHz, sensitive at 2 to 4 KHz.
• Dynamic range (quietest to loudest) is about 96 dB

Moreover, based on psycho-acoustic characteristics of human hearing, algorithms perform

some tricks to further reduce data rate
Masking Effects
• Frequency Masking: If a tone of a certain frequency and amplitude is
present, then other tones or noise of similar frequency cannot be
heard by the human ear
• the louder tone (masker) masks the softer tone (maskee)
• => no need to encode and transfer the softer tone
Masking Effects
• Repeat for various frequencies of masking tones
• Masking Threshold: Given a certain masker, the maximum non-
perceptible amplitude level of the softer tone. This threshold depends
upon the frequency, the type of masker, and the kind of sound being
masked.
Masking Effects
• Temporal Masking: If we hear a loud sound, then it stops, it takes a
little while until we can hear a soft tone nearby.
• The Masking Threshold is used by the audio encoder to determine the
maximum allowable quantization noise at each frequency to minimize
noise perceptibility: remove parts of signal that we cannot perceive
Speech Compression
• Handling speech with other media information such as text, images, video,
and data is the essential part of multimedia applications.
• The ideal speech coder has a low bit-rate, high perceived quality, low
signal delay, and low complexity.
• Delay
• Less than 150 ms one-way end-to-end delay for a conversation
• Processing (coding) delay, network delay
• Over Internet, ISDN, PSTN, ATM, …
• Complexity
• Computational complexity of speech coders depends on algorithms
• Contributes to achievable bit-rate and processing delay
G.72x Speech Coding Standards
• Quality
• – “intelligible” -> “natural” or “subjective” quality
• – Depending on bit-rate
• Bit-rate

MIPs: Million instructions per second

G.72x Audio Coding Standards
• Silence Compression - detect the "silence", similar to run-length coding
• Adaptive Differential Pulse Code Modulation (ADPCM)
• e.g., in CCITT G.721 -- 16 or 32 Kb/s.
• - (a) Encodes the difference between two or more consecutive signals; the
difference is then quantized
• -> hence the loss (speech quality becomes worse)
• (b) Adapts at quantization so fewer bits are used when the value is smaller.
• - It is necessary to predict where the waveform is headed -> difficult
• Linear Predictive Coding (LPC) fits signal to speech model and then
transmits parameters of model
• -> sounds like a computer talking, 2.4 Kb/s.
MPEG-1/2 Audio Compression
• 1. Use filters to divide the audio signal (e.g., 20-20kHz sound) into 32 frequency
subbands --> subband filtering.
• 2. Determine amount of masking for each band caused by nearby band using the
psycho-acoustic model.
• 3. If the power in a band is below masking threshold, don't encode it.
• 4. Otherwise, determine no. of bits needed to represent the coefficient such that
noise introduced by quantization is below the masking effect (one fewer bit of
quantization introduces about 6 dB of noise).
• 5. Format bitstream
MPEG Audio Compression Example
• After analysis, the first levels of 16 of the 32 bands are these:

• If the level of the 8th band is 60dB; it gives a masking (threshold) of 12 dB in the 7th band
and 15dB in the 9th.
• Level in 7th band is 10 dB ( < 12 dB ), so ignore it.
• Level in 9th band is 35 dB ( > 15 dB ), so send it.
• [ Only the amount above the masking (threshold) level needs to be sent, so instead of using
6 bits to encode it, we can use 4 bits – saving 2 bits (= 12 dB). ]
MPEG Audio Layers
• MPEG defines 3 layers for audio. Basic model is same, but codec complexity
increases with each layer.
• Divides data into frames, each of them contains 384 samples, 12 samples from each
of the 32 filtered subbands.
• Layer 1: DCT type filter with one frame and equal frequency spread per band.
Psycho-acoustic model only uses frequency masking.
• Layer 2: Use three frames in filter (before, current, next, a total of 1152 samples).
This models a little bit of the temporal masking.
• Layer 3: Better critical band filter is used (non-equal frequencies), psycho-acoustic
model includes temporal masking effects, takes into account stereo redundancy, and
uses Huffman coder
• MP3: Music compression format using MPEG Layer 3
 MPEG Audio Layers

Quality factor: 5 - perfect, 4 - just noticeable, 3 - slightly annoying, 2 - annoying, 1 - very annoying
• Real delay is about 3 times of the theoretical delay
MPEG-1 Audio Facts
• MPEG-1: 64K~320Kbps for audio
• – Uncompressed CD audio => 1.4 Mb/s
• • Compression factor ranging from 2.7 to 24.
• • With Compression rate 6:1 (16 bits stereo sampled at 48 KHz is reduced to 256 kb/s) and
optimal listening conditions, expert listeners could not distinguish between coded and
original audio clips.
• • MPEG audio supports sampling frequencies of 32, 44.1 and 48 KHz.
• • Supports one or two audio channels in one of the four modes:
• 1.Monophonic -- single audio channel
• 2.Dual-monophonic -- two independent chs, e.g., English and French
• 3.Stereo -- for stereo channels that share bits, but not using Jointstereo coding
• 4.Joint-stereo -- takes advantage of the correlations between stereo channels
MPEG-2 Audio Coding
• MPEG-2/MC: Provide theater-style surround sound capabilities
• - Five channels: left, right, center, rear left, and rear right
• – Five different modes: mono, stereo, three ch, four ch, five ch
• – Full five channel surround stereo: 640 Kb/s
• – 320 Kb/s for 5.1 stereo (5 channels+sub-woofer ch)
• • MPEG-2/LSF (Low sampling frequency: 16k, 22K, 24k)
• • MPEG-2/AAC (Advanced Audio Coding)
• - 7.1 channels
• - More complex coding
• • Compatibility:
• – Forward: MPEG-2 decoder can decode MPEG-1 bitstream
• – Backward: MPEG-1 decoder can decode a part of MPEG-2
MPEG-4 Audio Coding
• Consists of natural coding and synthetic coding
• • Natural coding
• - General coding: AAC and TwinVQ based arbitrary audio twice as good as MP3
• - Speech coding:
• * CELP I: 16K samp., 14.4~22.5Kbps
• * CELP II: 8K & 16K samp., 3.85~23.8Kbps
• * HVXV: 8M samp., 1.4~4Kbps
• • Synthetic coding: structured audio
• – Interface to Text-to-Speech synthesizers
• – High-quality audio synthesis with Structured Audio
• • AudioBIFS: Mix and postproduce multi-track sound streams
Structured Audio
• A description format that is made up of semantic information about the
sounds it represents, and that makes use of high-level (algorithmic) models.
• – E.g., MIDI (Musical Instrument Digital Interface).
• • Normal music digitization: perform waveform coding (we sample the music
signal and then try to reconstruct it exactly)
• • MIDI: only record musical actions such as the key depressed, the time when
the key is depressed, the duration for which the key remains depressed, and
how hard the key is struck (pressure).
• • MIDI is an example of parameter or event-list representation
• – An event list is a sequence of control parameters that, taken alone
• – Do not define the quality of a sound but instead specify the ordering and
characteristics of parts of a sound with regards to some external model.
Structured Audio Synthesis
• Sampling synthesis
• – Individual instrument sounds are digitally recorded and stored in memory
• – When the instrument is played, the note recording are reproduced and
mixed (added together) to produce the output sound.
• – This can be a very effective and realistic but requires a lot of memory
• – Good for playing music but not realistic for speech synthesis
• – Good for creating special sound effects from sample libraries
Structured Audio Synthesis
• Additive and subtractive synthesis
• – synthesize sound from the superposition of sinusoidal components
(additive)
• – Or from the filtering of an harmonically rich source sound - typically a
periodic oscillator with various form of waves (subtractive).
• – Very compact representation of the sound
• – the resulting notes often have a distinctive “analog synthesizer” character.
Applications of Structured Audio
• Low-bandwidth transmission
• – transmit a structural description and dynamically render it into sound on
the client side rather than rendering in a studio on the server side
• Sound generation from process models
• – the sound is not created from an event list but rather is dynamically
generated in response to evolving, nonsound-oriented environments such as
video games
• Music applications
• Content-based retrieval
• Virtual reality together with VRML/X3D
Common Audio File Formats
• Mulaw (Sun, NeXT) .au
• RIFF Wave (MS WAV) .wav
• MPEG Audio Layer (MPEG) .mp2 .mp3
• AIFC (Apple, SGI) .aiff .aif
• HCOM (Mac) .hcom
• SND (Sun, NeXT) .snd
• VOC (Soundblaster card proprietary standard) .voc
• AND MANY OTHERS!