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46-656: The Dolby TrueHD specification provides for up to 16 discrete audio channels, each with a sampling rate of up to 192 kHz and sample depth of up to 24 bits. Dolby's compression mechanism for TrueHD is Meridian Lossless Packing (MLP); prior to Dolby TrueHD, MLP was used for the DVD-Audio format, although the two formats' respective implementations of MLP are not mutually compatible. A Dolby TrueHD audio stream varies in bitrate , as does any other losslessly compressed audio format. Like its predecessor, Dolby TrueHD's bitstream carries program metadata , or non-audio information that

92-553: A bandpass signal is sampled slower than its Nyquist rate , the samples are indistinguishable from samples of a low-frequency alias of the high-frequency signal. That is often done purposefully in such a way that the lowest-frequency alias satisfies the Nyquist criterion , because the bandpass signal is still uniquely represented and recoverable. Such undersampling is also known as bandpass sampling , harmonic sampling , IF sampling , and direct IF to digital conversion. Oversampling

138-457: A moiré pattern . The process of volume rendering samples a 3D grid of voxels to produce 3D renderings of sliced (tomographic) data. The 3D grid is assumed to represent a continuous region of 3D space. Volume rendering is common in medical imaging, X-ray computed tomography (CT/CAT), magnetic resonance imaging (MRI), positron emission tomography (PET) are some examples. It is also used for seismic tomography and other applications. When

184-548: A 7.1 track to a 5.1 output, or a 5.1 track to a stereo output) by merging discrete channels' signals (except the low-frequency effects channel, the ".1," in a stereo mixdown, which is discarded due to its sound not playing back well without a dedicated subwoofer ). Dolby TrueHD is an optional codec, which means that Blu-ray hardware may decode it, but also may not (for example, inexpensive or early players, Blu-ray computer software, or pre–Blu-ray AV receivers). Consequently, all Blu-rays that include Dolby TrueHD audio also include

230-413: A Nyquist rate of B {\displaystyle B} , because all of its non-zero frequency content is shifted into the interval [ − B / 2 , B / 2 ] {\displaystyle [-B/2,B/2]} . Although complex-valued samples can be obtained as described above, they are also created by manipulating samples of a real-valued waveform. For instance,

276-482: A TrueHD bitstream, or more than two channels of PCM audio, using S/PDIF requires either falling back to a disc's Dolby Digital track or mixing the TrueHD track down to stereo. Sampling (signal processing)#Sampling rate In signal processing , sampling is the reduction of a continuous-time signal to a discrete-time signal . A common example is the conversion of a sound wave to a sequence of "samples". A sample

322-682: A decoder uses to modify its interpretation of the audio data. Dolby TrueHD metadata may include, for example, audio normalization or dynamic range compression . In addition, Dolby Atmos , a multi-dimensional surround format encoded using Dolby TrueHD, can embed more advanced metadata to spatially place sound objects in an Atmos-compatible speaker system. In the Blu-ray Disc specification, Dolby TrueHD tracks may carry up to 8 discrete audio channels ( 7.1 surround ) of 24-bit audio at 96 kHz, or up to 6 channels ( 5.1 surround ) at 192 kHz. The maximum bitrate of an audio stream including metadata

368-524: A digital low-pass filter whose cutoff frequency is B / 2 {\displaystyle B/2} . Computing only every other sample of the output sequence reduces the sample rate commensurate with the reduced Nyquist rate. The result is half as many complex-valued samples as the original number of real samples. No information is lost, and the original s ( t ) {\displaystyle s(t)} waveform can be recovered, if necessary. Signal processing Signal processing

414-582: A fail-safe track of Dolby Digital (AC-3), a mandatory codec. Unlike the competing DTS-HD Master Audio , which encodes its primary (optional) track in terms of differences from the companion mandatory track, a Dolby TrueHD-equipped Blu-ray's primary and companion tracks are redundant; the Dolby TrueHD bitstream has no data in common with the AC-3 bitstream, but AC-3 is used to construct E-AC3 stream. Similarly to DTS-HD MA, however, Dolby TrueHD's dual tracks are opaque to

460-417: A few GHz, and may be prohibitively expensive at much lower frequencies. Furthermore, while oversampling can reduce quantization error and non-linearity, it cannot eliminate these entirely. Consequently, practical ADCs at audio frequencies typically do not exhibit aliasing, aperture error, and are not limited by quantization error. Instead, analog noise dominates. At RF and microwave frequencies where oversampling

506-442: A measured signal. According to Alan V. Oppenheim and Ronald W. Schafer , the principles of signal processing can be found in the classical numerical analysis techniques of the 17th century. They further state that the digital refinement of these techniques can be found in the digital control systems of the 1940s and 1950s. In 1948, Claude Shannon wrote the influential paper " A Mathematical Theory of Communication " which

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552-427: A much lower rate. For most phonemes , almost all of the energy is contained in the 100 Hz – 4 kHz range, allowing a sampling rate of 8 kHz. This is the sampling rate used by nearly all telephony systems, which use the G.711 sampling and quantization specifications. Standard-definition television (SDTV) uses either 720 by 480 pixels (US NTSC 525-line) or 720 by 576 pixels (UK PAL 625-line) for

598-438: A proposed nonlinear function . Digital audio uses pulse-code modulation (PCM) and digital signals for sound reproduction. This includes analog-to-digital conversion (ADC), digital-to-analog conversion (DAC), storage, and transmission. In effect, the system commonly referred to as digital is in fact a discrete-time, discrete-level analog of a previous electrical analog. While modern systems can be quite subtle in their methods,

644-429: A sequence of samples, up to the Nyquist limit , by passing the sequence of samples through a reconstruction filter . Functions of space, time, or any other dimension can be sampled, and similarly in two or more dimensions. For functions that vary with time, let s ( t ) {\displaystyle s(t)} be a continuous function (or "signal") to be sampled, and let sampling be performed by measuring

690-597: A theoretical reconstruction is a common measure of the effectiveness of sampling. That fidelity is reduced when s ( t ) {\displaystyle s(t)} contains frequency components whose cycle length (period) is less than 2 sample intervals (see Aliasing ). The corresponding frequency limit, in cycles per second ( hertz ), is 0.5 {\displaystyle 0.5} cycle/sample × f s {\displaystyle f_{s}} samples/second = f s / 2 {\displaystyle f_{s}/2} , known as

736-430: Is 18 Mbit/s (instantaneous, since it is variable bitrate), and a TrueHD frame is either 1/1200 seconds long (for 48000 Hz, 96000 Hz or 192000 Hz) or 1/1102.5 seconds long (for 44100 Hz, 88200 Hz or 176400 Hz). Uncompressed (LPCM) it can be >35 Mbit/s. Any Blu-ray player or AV receiver that can decode TrueHD can also mix a multi-channel TrueHD track into any smaller amount of channels for final playback (for example,

782-530: Is a consequence of the Nyquist theorem . Sampling rates higher than about 50 kHz to 60 kHz cannot supply more usable information for human listeners. Early professional audio equipment manufacturers chose sampling rates in the region of 40 to 50 kHz for this reason. There has been an industry trend towards sampling rates well beyond the basic requirements: such as 96 kHz and even 192 kHz Even though ultrasonic frequencies are inaudible to humans, recording and mixing at higher sampling rates

828-450: Is a type of non-linear signal processing, where polynomial systems may be interpreted as conceptually straightforward extensions of linear systems to the non-linear case. Statistical signal processing is an approach which treats signals as stochastic processes , utilizing their statistical properties to perform signal processing tasks. Statistical techniques are widely used in signal processing applications. For example, one can model

874-434: Is a value of the signal at a point in time and/or space; this definition differs from the term's usage in statistics , which refers to a set of such values. A sampler is a subsystem or operation that extracts samples from a continuous signal . A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous signal at the desired points. The original signal can be reconstructed from

920-441: Is an electrical engineering subfield that focuses on analyzing, modifying and synthesizing signals , such as sound , images , potential fields , seismic signals , altimetry processing , and scientific measurements . Signal processing techniques are used to optimize transmissions, digital storage efficiency, correcting distorted signals, improve subjective video quality , and to detect or pinpoint components of interest in

966-502: Is converted to digital video , a different sampling process occurs, this time at the pixel frequency, corresponding to a spatial sampling rate along scan lines . A common pixel sampling rate is: Spatial sampling in the other direction is determined by the spacing of scan lines in the raster . The sampling rates and resolutions in both spatial directions can be measured in units of lines per picture height. Spatial aliasing of high-frequency luma or chroma video components shows up as

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1012-442: Is effective in eliminating the distortion that can be caused by foldback aliasing . Conversely, ultrasonic sounds may interact with and modulate the audible part of the frequency spectrum ( intermodulation distortion ), degrading the fidelity. One advantage of higher sampling rates is that they can relax the low-pass filter design requirements for ADCs and DACs , but with modern oversampling delta-sigma-converters this advantage

1058-574: Is either Analog signal processing is for signals that have not been digitized, as in most 20th-century radio , telephone, and television systems. This involves linear electronic circuits as well as nonlinear ones. The former are, for instance, passive filters , active filters , additive mixers , integrators , and delay lines . Nonlinear circuits include compandors , multipliers ( frequency mixers , voltage-controlled amplifiers ), voltage-controlled filters , voltage-controlled oscillators , and phase-locked loops . Continuous-time signal processing

1104-421: Is for sampled signals, defined only at discrete points in time, and as such are quantized in time, but not in magnitude. Analog discrete-time signal processing is a technology based on electronic devices such as sample and hold circuits, analog time-division multiplexers , analog delay lines and analog feedback shift registers . This technology was a predecessor of digital signal processing (see below), and

1150-489: Is for signals that vary with the change of continuous domain (without considering some individual interrupted points). The methods of signal processing include time domain , frequency domain , and complex frequency domain . This technology mainly discusses the modeling of a linear time-invariant continuous system, integral of the system's zero-state response, setting up system function and the continuous time filtering of deterministic signals Discrete-time signal processing

1196-435: Is impractical and filters are expensive, aperture error, quantization error and aliasing can be significant limitations. Jitter, noise, and quantization are often analyzed by modeling them as random errors added to the sample values. Integration and zero-order hold effects can be analyzed as a form of low-pass filtering . The non-linearities of either ADC or DAC are analyzed by replacing the ideal linear function mapping with

1242-431: Is less important. The Audio Engineering Society recommends 48 kHz sampling rate for most applications but gives recognition to 44.1 kHz for CD and other consumer uses, 32 kHz for transmission-related applications, and 96 kHz for higher bandwidth or relaxed anti-aliasing filtering . Both Lavry Engineering and J. Robert Stuart state that the ideal sampling rate would be about 60 kHz, but since this

1288-435: Is not a standard frequency, recommend 88.2 or 96 kHz for recording purposes. A more complete list of common audio sample rates is: Audio is typically recorded at 8-, 16-, and 24-bit depth, which yield a theoretical maximum signal-to-quantization-noise ratio (SQNR) for a pure sine wave of, approximately, 49.93  dB , 98.09 dB and 122.17 dB. CD quality audio uses 16-bit samples. Thermal noise limits

1334-444: Is sampled using an analog-to-digital converter (ADC), a device with various physical limitations. This results in deviations from the theoretically perfect reconstruction, collectively referred to as distortion . Various types of distortion can occur, including: Although the use of oversampling can completely eliminate aperture error and aliasing by shifting them out of the passband, this technique cannot be practically used above

1380-469: Is still more common on titles with non-Atmos lossless audio. Regardless, publishers such as Paramount Home Entertainment , and Crunchyroll still use Dolby TrueHD for their releases. Universal Pictures Home Entertainment has recently used Dolby TrueHD on occasion. Audio encoded using Dolby TrueHD may be transported to A/V receivers in one of three ways depending on player and/or receiver support: Because S/PDIF does not have sufficient bandwidth to carry

1426-705: Is still used in advanced processing of gigahertz signals. The concept of discrete-time signal processing also refers to a theoretical discipline that establishes a mathematical basis for digital signal processing, without taking quantization error into consideration. Digital signal processing is the processing of digitized discrete-time sampled signals. Processing is done by general-purpose computers or by digital circuits such as ASICs , field-programmable gate arrays or specialized digital signal processors . Typical arithmetical operations include fixed-point and floating-point , real-valued and complex-valued, multiplication and addition. Other typical operations supported by

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1472-467: Is the Hilbert transform of the other waveform, s ( t ) {\displaystyle s(t)} , the complex-valued function, s a ( t ) ≜ s ( t ) + i ⋅ s ^ ( t ) {\displaystyle s_{a}(t)\triangleq s(t)+i\cdot {\hat {s}}(t)} , is called an analytic signal , whose Fourier transform

1518-610: Is used in most modern analog-to-digital converters to reduce the distortion introduced by practical digital-to-analog converters , such as a zero-order hold instead of idealizations like the Whittaker–Shannon interpolation formula . Complex sampling (or I/Q sampling ) is the simultaneous sampling of two different, but related, waveforms, resulting in pairs of samples that are subsequently treated as complex numbers . When one waveform, s ^ ( t ) {\displaystyle {\hat {s}}(t)} ,

1564-519: Is zero for all negative values of frequency. In that case, the Nyquist rate for a waveform with no frequencies ≥  B can be reduced to just B (complex samples/sec), instead of 2 B {\displaystyle 2B} (real samples/sec). More apparently, the equivalent baseband waveform , s a ( t ) ⋅ e − i 2 π B 2 t {\displaystyle s_{a}(t)\cdot e^{-i2\pi {\frac {B}{2}}t}} , also has

1610-465: The Nyquist frequency of the sampler. Therefore, s ( t ) {\displaystyle s(t)} is usually the output of a low-pass filter , functionally known as an anti-aliasing filter . Without an anti-aliasing filter, frequencies higher than the Nyquist frequency will influence the samples in a way that is misinterpreted by the interpolation process. In practice, the continuous signal

1656-445: The equivalent baseband waveform can be created without explicitly computing s ^ ( t ) {\displaystyle {\hat {s}}(t)} , by processing the product sequence, [ s ( n T ) ⋅ e − i 2 π B 2 T n ] {\displaystyle \left[s(nT)\cdot e^{-i2\pi {\frac {B}{2}}Tn}\right]} , through

1702-708: The hardware are circular buffers and lookup tables . Examples of algorithms are the fast Fourier transform (FFT), finite impulse response (FIR) filter, Infinite impulse response (IIR) filter, and adaptive filters such as the Wiener and Kalman filters . Nonlinear signal processing involves the analysis and processing of signals produced from nonlinear systems and can be in the time, frequency , or spatiotemporal domains. Nonlinear systems can produce highly complex behaviors including bifurcations , chaos , harmonics , and subharmonics which cannot be produced or analyzed using linear methods. Polynomial signal processing

1748-400: The integration period may be significantly shorter than the time between repetitions, the sampling frequency can be different from the inverse of the sample time: Video digital-to-analog converters operate in the megahertz range (from ~3 MHz for low quality composite video scalers in early games consoles, to 250 MHz or more for the highest-resolution VGA output). When analog video

1794-443: The primary usefulness of a digital system is the ability to store, retrieve and transmit signals without any loss of quality. When it is necessary to capture audio covering the entire 20–20,000 Hz range of human hearing such as when recording music or many types of acoustic events, audio waveforms are typically sampled at 44.1 kHz ( CD ), 48 kHz, 88.2 kHz, or 96 kHz. The approximately double-rate requirement

1840-517: The sample values. When the time interval between adjacent samples is a constant ( T ) {\displaystyle (T)} , the sequence of delta functions is called a Dirac comb . Mathematically, the modulated Dirac comb is equivalent to the product of the comb function with s ( t ) {\displaystyle s(t)} . That mathematical abstraction is sometimes referred to as impulse sampling . Most sampled signals are not simply stored and reconstructed. The fidelity of

1886-455: The true number of bits that can be used in quantization. Few analog systems have signal to noise ratios (SNR) exceeding 120 dB. However, digital signal processing operations can have very high dynamic range, consequently it is common to perform mixing and mastering operations at 32-bit precision and then convert to 16- or 24-bit for distribution. Speech signals, i.e., signals intended to carry only human speech , can usually be sampled at

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1932-402: The unit samples per second , sometimes referred to as hertz , for example 48 kHz is 48,000 samples per second . Reconstructing a continuous function from samples is done by interpolation algorithms. The Whittaker–Shannon interpolation formula is mathematically equivalent to an ideal low-pass filter whose input is a sequence of Dirac delta functions that are modulated (multiplied) by

1978-482: The user; a Blu-ray player loaded with a Dolby TrueHD disc will automatically fall back to AC-3 if it cannot decode or pass through the lossless bitstream, with no explicit selection required (or offered). Dolby TrueHD's prominence relative to DTS-HD MA began to decline around 2010. It has experienced a mild resurgence as the encoding used for Dolby Atmos audio (especially in Ultra HD Blu-ray titles), but DTS-HD MA

2024-484: The value of the continuous function every T {\displaystyle T} seconds, which is called the sampling interval or sampling period . Then the sampled function is given by the sequence: The sampling frequency or sampling rate , f s {\displaystyle f_{s}} , is the average number of samples obtained in one second, thus f s = 1 / T {\displaystyle f_{s}=1/T} , with

2070-435: The visible picture area. High-definition television (HDTV) uses 720p (progressive), 1080i (interlaced), and 1080p (progressive, also known as Full-HD). In digital video , the temporal sampling rate is defined as the frame rate  – or rather the field rate  – rather than the notional pixel clock . The image sampling frequency is the repetition rate of the sensor integration period. Since

2116-560: Was published in the Bell System Technical Journal . The paper laid the groundwork for later development of information communication systems and the processing of signals for transmission. Signal processing matured and flourished in the 1960s and 1970s, and digital signal processing became widely used with specialized digital signal processor chips in the 1980s. A signal is a function x ( t ) {\displaystyle x(t)} , where this function

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