KEN POHLMANN PRINCIPLES OF DIGITAL AUDIO PDF

Technology Nonfiction The definitive guide to digital engineering—fully updated Gain a thorough understanding of digital audio tools, techniques, and practices from this completely revised and expanded resource. Covering basic theory to the latest technological advancements, the book explains how to apply digital conversion, processing, compression, storage, streaming, and transmission concepts. New chapters on Blu-ray, speech coding, and low bit-rate coding are also included in this bestselling guide. Learn about discrete time sampling, quantization, and signal processing Examine details of CD, DVD, and Blu-ray players and discs Encode and decode AAC, MP3, MP4, Dolby Digital, and other files Prepare content for distribution via the Internet and digital radio and television Learn the critical differences between music coding and speech coding Design low bit-rate codecs to optimize memory capacity while preserving fidelity Develop methodologies to evaluate the sound quality of music and speech files Study audio transmission via HDMI, VoIP, Wi-Fi, and Bluetooth Handle digital rights management, fingerprinting, and watermarking Understand how one-bit conversion and high-order noise shaping work.

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Pohlmann and Other Sources Summary by Michael McGoodwin, prepared I prepared this summary primarily for my own benefit to improve my understanding of these important technologies. You may find it useful particularly in searching for relevant technical terms and acronyms in order to find their meaning in context.

I have not attempted to prevent a comprehensive summary—only a listing of selected key, confusing, or complex points, specifications, design features and shortcomings, etc. I would be pleased to be notified of any errors. I highly recommend the Pohlmann text to any technically-minded individual interested in improving his or her knowledge about these current and emerging technologies. Although the emphasis of the book is on audio technologies, there is much useful information also about video, cinefilm audio, DVD Digital Versatile Disk , and other digital technologies.

Page and chapter references are to that text unless otherwise indicated. I have attempted to translate measurements to a uniform set of units and have corrected several errors found in the text. Binary coded decimal BCD " code" in which each decimal digit is coded as 4 binary bits having weights or values of 8, 4, 2 and 1. Chapter 2: Fundamentals of Digital Audio [Review of basic overall principles] In analog recording systems, the continuously varying amplitude of the sound waveform is translated to a continuously varying level of magnetism, LP groove amplitude, etc.

In contrast, with digital audio encoding, discrete noncontinuous time sampling and amplitude quantization is required, a process that breaks the originally continuous and smooth waveform into a staircase or other pattern of pulses.

Sampling: Sampling theorem of Harry Nyquist and his precursors states that a continuous band-limited signal may be exactly represented without any loss of data by amplitude samples made at a sampling frequency S equal to twice the highest signal frequency component. The reconstruction of the original input analog signal is accomplished by an output lowpass filter which interpolates the staircase waveform pattern of the digital signal.

Sampling frequency determines the audio bandwidth of the system. Prevention of aliasing requires the analog input signal to be passed through an analog lowpass antialiasing filter before sampling to insure that no higher frequencies are present. Digitization can never perfectly encode a signal, it is only an approximation due to measurement error and moreover, the precise capabilities of human hearing are not known. The percentage of distortion increases as signal strength decreases Low level signal quantization can induce aliased components "granulation noise" and beat tones or "birdies" which may be unpleasantly audible.

Dither must be added to prevent these problems or sigma delta modulation and noise shaping used. Dithering adds, prior to sampling, a small amount of noise that is uncorrelated with the signal.

This increases total noise in the form of white noise but reduces distortion. Dither effectively permits encoding below the LSB level e. The word dither derives from ME "didderen" meaning "to tremble"—it was found that WWII airplane bomb sights perfomed better when vibrating than when still, similar to the improvement in accuracy gained from tapping a meter. A typical recording system includes input amplifier, dither generator, input lowpass filter, sample-and-hold circuit, analog to digital converter ADC newer generation designs usually combine mild lowpass filtering, oversampling, and decimation processing , multiplexer to combine channels , digital processing and modulation circuit channel coding , and a storage medium.

Filters may introduce phase distortion. Many are described by their mathematical polynomials: Bessel, Butterworth, Chebyshev, etc.

Filters may have several cascaded stages or orders which improve the approximation to brick-wall filtering but worsen phase shift. No filter is ideal in all respects. Oversampling and sigma delta conversion allows a more gradual cutoff frequency and has mostly replaced analog brick wall filtering. Its output is a discrete PAM staircase i. Any variation in absolute timing comprises "jitter", which adds noise and distortion.

Jitter is worst for high amplitude high frequency signals. Jitter must be less than ps picoseconds for 16 bit sampling of a 20 kHz full amplitude sinewave and less than ps for 20 bit in order to keep resultant noise below the quantization noise floor.

Problems also arise if the hold signal "droops" before it is read due to capacitor leakage etc the value must be held to within 1 mV during conversion.

Typical circuits employ a junction field effect transistor JFET operational amplifier combined with a capacitor and require accurate high speed switching and acquisition time.

The analog to digital converter ADC is the most critical component. It must have a precision of 15 parts per million for 16 bit intervals and 1 ppm for 20 bit. Speed and accuracy are required. Conversion time must be less than one sampling period T. Settling time or propagation errors can occur. Integral monotonic linearity is required—i. Some drift with temperature may occur and a regulated power supply is needed. Code width "quantum" is the range of analog input signal over which a given output value will occur—ideally it should be 1 LSB Despite this limitation, internal processing may require longer word lengths.

Oversampling uses a high oversampling rate R say of 72 e. They obviate brick wall filters and provide increased resolution. Sigma delta modulation conversion SDM is employed and a decimation filter to downsample i. Also multiplexing to combine multiple channels in a single serial stream , adding redundant parity data for error detection and correction, perform interleaving to spread the data widely in the bitstream in order to improve error recoverability , grouping data into frames which include synchronization and data headers [e.

Channel coding channel modulation creates the actual modulated signal channel code which is stored on the media. Channel codes for audio are ususally combined with a clock pulse to make the encoded waveform self-clocking i. Tminis the minimum allowable time interval between transitions i.

It determines the highest frequency that can be stored higher Tmin allows higher maximum frequency. Code data storage efficiency decreases with the addition of increasingly accurate and therefore frequent clocking data so the relative requirement for each must be weighed.

The density ratio DR is the ratio between Tmin and the length of a single bit period T the shortest time interval between transitions of input data to be modulated , i. The Tmax is the longest interval of time between transitions allowed in the modulated signal—increasing clocking accuracy lowers Tmax. The window margin Tw phase margin or jitter margin reflects the tolerance for errors in locating the transition or resistance to jitter, higher is better.

For efficiency, transitions between values are usually stored rather than absolute PCM values. Coding must restrict DC content in many applications which can be monitored by the digital sum value DSV. Group codes represent a type of run length limited RLL coding and therefore the spacing of transitions may be a multiple of T. Chapter 4: Digital Audio Reproduction [Overview of steps involved in digital reproduction or playback] Reproduction i.

Problems include differential nonlinearity wide or narrow codes causing missing codes, which are worst for low-level signals and can result in audible harmonic and intermodulation distortion and nonmonotonicity. Testing should use test frequencies that are not correlated with the sampling freq. Zero-cross distortion arising where the signal goes from positive to negative can be audible and is aggravated by dithering. Some manufacturers use or bit conversion to convert bit PCM code in order to improve fidelity, reasoning that signal digitization and processing steps should have a greater dynamic range than the final recording.

Oversampling does not ultimately create new information—it makes better use of existing information. DAC low level distortion is best measured by the dB ratio of maximal signal to the broadband noise 0 to 20 kHz when reproducing a dB signal—this can allow the computation of the effective number of bits ENOB, which might be In practice today, sigma-delta converters and high oversampling rates are used.

The latter results in attentuation of high frequencies and arises from the finite and narrow pulse width. Output Lowpass Anti-imaging or smoothing Filter: This stage is analogous to the input anti-aliasing filter used with recording [but has been supplanted by digital filtering prior to DAC].

This filter must remove all frequency content above the half-sampling frequency, thereby converting the now analog PAM staircase waveform into a smoothly continuous final waveform. These must have a flat passband and highly attenutated stopband including even the extreme high frequencies, where artefacts arising from digital processing are found , with a steep cutoff slope. Phase shifts must be minimized or corrected.

Oversampling techniques have eliminated the need for brick-wall filters. Even though stopband frequencies for high-quality audio systems are inaudible, they must still be filtered out to prevent downstream "modulation" into audible frequencies. It can be characterized in the time domain or frequency domain, related through the Fourier transform. An ideal brick wall filter in the frequency domain [i. Digital Filter: These are used prior to DA conversion and have supplanted analog brick-wall filters.

Usually these are finite impulse response FIR filters and can be simple in design when oversampling is used. Spectral images sidebands can be removed with a gentle analog filter because they do not appear except at a high frequency exceeding the high oversampling cutoff frequency fa. A typical diigital filter utilizes a digital transversal filter with tapped delay lines, multipliers, and an adder. Digital filters are not affected by temperature.

The 12 least significant bits are typically delayed 1 sampling period and subtracted from the next data word. This noise shaping decreases the noise floor by 7 dB in the audio band though it increases noise in higher bands. Alternate coding and quantizing methods: Though most systems use linear PCM LPCM coding, some use companding dynamic compression and decompression employing many differing quantization intervals in the low amplitudes versus the high amplitudes.

These may be floating point or logarithmic etc. Block floating point systems allow data reduction. Differential PCM Delta modulation is a true 1-bit method using a very high sampling frequency but fails to perform well in high fidelity applications though sigma-delta modulation works very well and is used with the SACD. Any variation can be characterized as jitter.

Jitter manifests as broadband noise from random jitter or a single spectral line from periodic jitter. The oscilloscopic eye pattern displays success transitions and can demonstrate jitter. Jitter can arise from the interface as well as from sampling, and in the storage media itself Sigma delta converters can be very sensitive to jitter—it is not usually a problem in well designed audio equipment.

Chapter 5: Error Correction [Mathematical and Physical Technologies for Error Detection, Correction, and Prevention] Errors are inevitable but by means of robust error correction systems, CD and DVD can have uncorrectable error rates as low as that specified for computers, i.

Audio applications do not require this degree of accuracy. Measures of error: The burst length is the maximum number of adjacent erroneous bits that can be fully corrected. The bit-error rate BER is the number of error bits per total bits. Optical disk systems can handle BERs of to The block error rate BLER is the rate of block or frames per second having at least one incorrect bit.

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