In this presentation, production of digital audio is discussed. Also brief introduction about digital audio broadcast, recording techniques and stereo phony is given.
Transcript of Digital audio
Digital Audio
Contents Digital Audio Fundamentals Sampling and Quantizing PCM
Audio Compression Disk-Based Recording Rotary Head Digital
Recorders Digital Audio Broadcasting Digital Filtering Stereophony
and Multichannel Sound
Digital Audio Fundamentals Digital audio is simply an
alternative means of carrying an audio waveform
Digital Audio Fundamentals Digital audio is sound reproduction
using pulse-code modulation and digital signals Digital audio
systems include analog-to-digital conversion (ADC),
digital-to-analog conversion (DAC), digital storage, processing and
transmission components A primary benefit of digital audio is in
its convenience of storage, transmission and retrieval Digital
audio is useful in the recording, manipulation, mass-production,
and distribution of sound Modern distribution of music across the
Internet via on-line stores depends on digital recording and
digital compression algorithms
PCM(Pulse Code Modulation) PCM consists of three steps to
digitize an analog signal: 1. Sampling 2. Quantization 3. Binary
encoding Before we sample, we have to filter the signal to limit
the maximum frequency of the signal as it affects the sampling
rate. Filtering should ensure that we do not distort the signal, ie
remove high frequency components that affect the signal shape.
PCM(Pulse Code Modulation)
Sampling Analog signal is sampled every TS secs. Ts is referred
to as the sampling interval. fs = 1/Ts is called the sampling rate
or sampling frequency. There are 3 sampling methods: Ideal - an
impulse at each sampling instant Natural - a pulse of short width
with varying amplitude Flattop - sample and hold, like natural but
with single amplitude value The process is referred to as pulse
amplitude modulation PAM and the outcome is a signal with analog
(non integer) values
Sampling
Sampling
Quantization Sampling results in a series of pulses of varying
amplitude values ranging between two limits: a min and a max The
amplitude values are infinite between the two limits. We need to
map the infinite amplitude values onto a finite set of known values
This is achieved by dividing the distance between min and max into
L zones, each of height = (max - min)/L The midpoint of each zone
is assigned a value from 0 to L-1 (resulting in L values) Each
sample falling in a zone is then approximated to the value of the
midpoint
Quantization Zones Assume we have a voltage signal with
amplitutes Vmin=-20V and Vmax=+20V. We want to use L=8 quantization
levels. Zone width = (20 - -20)/8 = 5 The 8 zones are: -20 to -15,
-15 to -10, -10 to -5, -5 to 0, 0 to +5, +5 to +10, +10 to +15, +15
to +20 The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5,
17.5
Assigning Codes to Zones Each zone is then assigned a binary
code. The number of bits required to encode the zones, or the
number of bits per sample as it is commonly referred to, is
obtained as follows: nb = log2 L Given our example, nb = 3 The 8
zone (or level) codes are therefore: 000, 001, 010, 011, 100, 101,
110, and 111 Assigning codes to zones: 000 will refer to zone -20
to -15 001 to zone -15 to -10, etc.
Quantization and encoding of a sampled signal
PCM Decoder To recover an analog signal from a digitized signal
we follow the following steps: We use a hold circuit that holds the
amplitude value of a pulse till the next pulse arrives. We pass
this signal through a low pass filter with a cutoff frequency that
is equal to the highest frequency in the pre-sampled signal. The
higher the value of L, the less distorted a signal is
recovered.
PCM Decoder
Audio Compression In its native form, high-quality digital
audio requires a high data rate, which may be excessive for certain
applications One approach to the problem is to use compression,
which reduces that rate significantly with a moderate loss of
subjective quality While compression may achieve considerable
reduction in bit rate, it must be appreciated that compression
systems reintroduce the generation loss of the analog domain to
digital systems
Audio Compression One of the most popular compression standards
for audio and video is known as MPEG (Moving Picture Experts Group)
In practice, audio and video streams of this type can be combined
using multiplexing The program stream is optimized for recording
and is based on blocks of arbitrary size The transport stream is
optimized for transmission and is based on blocks of constant
size
Audio Compression The bit stream types of MPEG-2
Audio Compression Compression and the corresponding decoding
are complex processes and take time, adding to existing delays in
signal paths Concealment of uncorrectable errors is also more
difficult on compressed data The acceptable trade-off between loss
of audio quality and transmission or storage size depends upon the
application For example, one 640MB compact disc (CD) holds
approximately one hour of uncompressed high fidelity music, less
than 2 hours of music compressed losslessly, or 7 hours of music
compressed in the MP3 format at a medium bit rate A digital sound
recorder can typically store around 200 hours of clearly
intelligible speech in 640MB
Disk-Based Recording The magnetic disk drive was perfected by
the computer industry to allow rapid random access to data, and so
it makes an ideal medium for editing Development of the optical
disk was stimulated by the availability of low-cost lasers Optical
disks are available in many different types, some which can only be
recorded once, whereas others are erasable Optical disks have in
common the fact that access is generally slower than with magnetic
drives and that it is difficult to obtain high data rates, but most
of them are removable and can act as interchange media
Rotary Head Digital Recorders In a fixed tape head system,
audio tape is drawn past the head at a constant speed The head
creates a fluctuating magnetic field in response to the signal to
be recorded, and the magnetic particles on the tape are forced to
line up with the field at the head As the tape moves away, the
magnetic particles carry an imprint of the signal in their magnetic
orientation If the tape moves too slowly, a high frequency signal
will not be imprinted: the particles' polarity will simply
oscillate in the vicinity of the head, to be left in a random
position
Rotary Head Digital Recorders Thus the bandwidth channel
capacity of the recorded signal can be seen to be related to tape
speed: the faster the speed, the higher the frequency that can be
recorded Digital video and digital audio need considerably more
bandwidth than analog audio, so much so that tape would have to be
drawn past the heads at very high speed in order to capture this
signal This is impractical, since tapes of immense length would be
required The generally adopted solution is to rotate the head
against the tape at high speed, so that the relative velocity is
high, but the tape itself moves at a slow speed.
Rotary Head Digital Recorders To accomplish this, the head must
be tilted so that at each rotation of the head, a new area of tape
is brought into play; each segment of the signal is recorded as a
diagonal stripe across the tape This is known as a helical scan
because the tape wraps around the circular drum at an angle,
travelling up like a helix The rotary head recorder has the
advantage that the spinning heads create a high head-to-tape speed,
offering a high bit rate recording without high linear tape
speed
Digital Audio Broadcasting Digital Audio Broadcasting (DAB) is
a digital radio technology for broadcasting radio stations
Advantages of DAB Broadcasting programs with good sound quality
comparable to multi-media products such as MP3 Offering stable
reception and removing noises Bringing diversified program choices
to the audiences Enabling transmission of text / images
DAB sender Trans- mitter Trans- mission Multi- plexer MSC
Multi- plexer ODFM Packet Mux Channel Coder Audio Encoder Channel
Coder DAB Signal Service Information FIC Multiplex Information Data
Services Audio Services Radio Frequency FIC: Fast Information
Channel MSC: Main Service Channel OFDM: Orthogonal Frequency
Division Multiplexing 1.5 MHz f carriers
DAB receiver Packet Demux Audio Decoder Channel Decoder
Independent Data Service Audio Service Controller Tuner ODFM
Demodulator User Interface FIC Control Bus (partial) MSC
Digital Filtering In electronics, computer science and
mathematics, a digital filter is a system that performs
mathematical operations on a sampled, discrete- time signal to
reduce or enhance certain aspects of that signal A digital filter
system usually consists of an analog-to-digital converter to sample
the input signal, followed by a microprocessor and some peripheral
components such as memory to store data and filter
coefficients
Digital Filtering Digital filters are commonplace and an
essential element of everyday electronics such as radios,
cellphones, and stereo receivers Digital filters are defined by
their impulse response, h[n], or the filter output given a unit
sample impulse input signal A discrete-time unit impulse signal is
defined by
Digital Filtering Digital filters are often best described in
terms of their frequency response. That is, how is a sinusoidal
signal of a given frequency affected by the filter The frequency
response of a digital filter can be found by taking the DFT (or
FFT) of the filter impulse response The frequency response of a
filter consists of its magnitude and phase responses The magnitude
response indicates the ratio of a filtered sine wave's output
amplitude to its input amplitude The phase response describes the
phase ``offset'' or time delay experienced by a sine wave passing
through a filter
Digital Filtering The filter implementation simply performs a
convolution of the time domain impulse response and the sampled
signal Convolution is defined as the integral of the product of the
two functions after one is reversed and shifted or delayed What
happens when we add a signal to a one-sample delayed version of
itself? y[n] = x[n] + x[n - 1]
Digital Filtering Consider the following input signals: The
filter's frequency response magnitude is shown
Digital Filtering Finite Impulse Response (FIR) Filters Finite
Impulse Response (FIR) filters are defined by scaled and
time-delayed versions of the filter input signal only, as given by
the following difference equation: The impulse response of an FIR
filter is only as long as the maximum delayed input term in its
difference equation
Digital Filtering An FIR filter can be represented by a block
diagram as shown
Digital Filtering What happens if we use a previous filter
output value to produce the filter's current output? y[n] = x[n] +
y[n - 1] Consider the following input signals
Digital Filtering The filter's frequency response magnitude is
shown
Digital Filtering Infinite Impulse Response (IIR) filters
include delayed and scaled versions of the output signal which are
fed back into the current output IIR filters are described by the
following difference equation
Digital Filtering An IIR filter can be represented by a block
diagram as shown
Stereophony and Multichannel Sound It is a method of sound
recording in which the recording contains information about the
spatial arrangement of the sound sources When a stereophonic
recording is reproduced, the listener hears a more natural sound
that seems to come from many separate sources and to be arranged in
the same way as during the recording The listener has the
impression that the sound is three-dimensional and possessed of an
added depth.
Stereophony and Multichannel Sound This effect is achieved
through the separate recording of electrical signals from different
microphones on individual channels and through the separate
reproduction of the sound on each channel by loudspeakers
Stereophony and Multichannel Sound The arrangement of the
loudspeakers must be similar to that of the microphones; that is,
the right and left channels must coincide The quality of
stereophonic sound reproduction improves with the number of
channels used However, the number of channels is usually kept
within certain limits to avoid undue complexity and excessive
cost
Stereophony and Multichannel Sound 5.1 channel sound is an
industry standard sound format for movies and music with five main
channels of sound and a sixth subwoofer channel used for special
movie effects and bass for music A 5.1 channel system consists of a
stereo pair of speakers, a center channel speaker placed between
the stereo speakers and two surround sound speakers located behind
the listener. 5.1 channel sound is found on DVD movie and music
discs and some CDs
Stereophony and Multichannel Sound 6.1 channel sound is a sound
enhancement to 5.1 channel sound with an additional center surround
sound speaker located between the two surround sound speakers
directly behind the listener. 6.1 channel sound produces a more
enveloping surround sound experience. 7.1 channel sound is a
further sound enhancement to 5.1 channel sound with two additional
side-surround speakers located to the sides of the listeners
seating position. 7.1 channel sound is used for greater sound
envelopment and more accurate positioning of sounds