
- •Preface
- •Introduction
- •1.1 Spatial coordinate systems
- •1.2 Sound fields and their physical characteristics
- •1.2.1 Free-field and sound waves generated by simple sound sources
- •1.2.2 Reflections from boundaries
- •1.2.3 Directivity of sound source radiation
- •1.2.4 Statistical analysis of acoustics in an enclosed space
- •1.2.5 Principle of sound receivers
- •1.3 Auditory system and perception
- •1.3.1 Auditory system and its functions
- •1.3.2 Hearing threshold and loudness
- •1.3.3 Masking
- •1.3.4 Critical band and auditory filter
- •1.4 Artificial head models and binaural signals
- •1.4.1 Artificial head models
- •1.4.2 Binaural signals and head-related transfer functions
- •1.5 Outline of spatial hearing
- •1.6 Localization cues for a single sound source
- •1.6.1 Interaural time difference
- •1.6.2 Interaural level difference
- •1.6.3 Cone of confusion and head movement
- •1.6.4 Spectral cues
- •1.6.5 Discussion on directional localization cues
- •1.6.6 Auditory distance perception
- •1.7 Summing localization and spatial hearing with multiple sources
- •1.7.1 Summing localization with two sound sources
- •1.7.2 The precedence effect
- •1.7.3 Spatial auditory perceptions with partially correlated and uncorrelated source signals
- •1.7.4 Auditory scene analysis and spatial hearing
- •1.7.5 Cocktail party effect
- •1.8 Room reflections and auditory spatial impression
- •1.8.1 Auditory spatial impression
- •1.8.2 Sound field-related measures and auditory spatial impression
- •1.8.3 Binaural-related measures and auditory spatial impression
- •1.9.1 Basic principle of spatial sound
- •1.9.2 Classification of spatial sound
- •1.9.3 Developments and applications of spatial sound
- •1.10 Summary
- •2.1 Basic principle of a two-channel stereophonic sound
- •2.1.1 Interchannel level difference and summing localization equation
- •2.1.2 Effect of frequency
- •2.1.3 Effect of interchannel phase difference
- •2.1.4 Virtual source created by interchannel time difference
- •2.1.5 Limitation of two-channel stereophonic sound
- •2.2.1 XY microphone pair
- •2.2.2 MS transformation and the MS microphone pair
- •2.2.3 Spaced microphone technique
- •2.2.4 Near-coincident microphone technique
- •2.2.5 Spot microphone and pan-pot technique
- •2.2.6 Discussion on microphone and signal simulation techniques for two-channel stereophonic sound
- •2.3 Upmixing and downmixing between two-channel stereophonic and mono signals
- •2.4 Two-channel stereophonic reproduction
- •2.4.1 Standard loudspeaker configuration of two-channel stereophonic sound
- •2.4.2 Influence of front-back deviation of the head
- •2.5 Summary
- •3.1 Physical and psychoacoustic principles of multichannel surround sound
- •3.2 Summing localization in multichannel horizontal surround sound
- •3.2.1 Summing localization equations for multiple horizontal loudspeakers
- •3.2.2 Analysis of the velocity and energy localization vectors of the superposed sound field
- •3.2.3 Discussion on horizontal summing localization equations
- •3.3 Multiple loudspeakers with partly correlated and low-correlated signals
- •3.4 Summary
- •4.1 Discrete quadraphone
- •4.1.1 Outline of the quadraphone
- •4.1.2 Discrete quadraphone with pair-wise amplitude panning
- •4.1.3 Discrete quadraphone with the first-order sound field signal mixing
- •4.1.4 Some discussions on discrete quadraphones
- •4.2 Other horizontal surround sounds with regular loudspeaker configurations
- •4.2.1 Six-channel reproduction with pair-wise amplitude panning
- •4.2.2 The first-order sound field signal mixing and reproduction with M ≥ 3 loudspeakers
- •4.3 Transformation of horizontal sound field signals and Ambisonics
- •4.3.1 Transformation of the first-order horizontal sound field signals
- •4.3.2 The first-order horizontal Ambisonics
- •4.3.3 The higher-order horizontal Ambisonics
- •4.3.4 Discussion and implementation of the horizontal Ambisonics
- •4.4 Summary
- •5.1 Outline of surround sounds with accompanying picture and general uses
- •5.2 5.1-Channel surround sound and its signal mixing analysis
- •5.2.1 Outline of 5.1-channel surround sound
- •5.2.2 Pair-wise amplitude panning for 5.1-channel surround sound
- •5.2.3 Global Ambisonic-like signal mixing for 5.1-channel sound
- •5.2.4 Optimization of three frontal loudspeaker signals and local Ambisonic-like signal mixing
- •5.2.5 Time panning for 5.1-channel surround sound
- •5.3 Other multichannel horizontal surround sounds
- •5.4 Low-frequency effect channel
- •5.5 Summary
- •6.1 Summing localization in multichannel spatial surround sound
- •6.1.1 Summing localization equations for spatial multiple loudspeaker configurations
- •6.1.2 Velocity and energy localization vector analysis for multichannel spatial surround sound
- •6.1.3 Discussion on spatial summing localization equations
- •6.1.4 Relationship with the horizontal summing localization equations
- •6.2 Signal mixing methods for a pair of vertical loudspeakers in the median and sagittal plane
- •6.3 Vector base amplitude panning
- •6.4 Spatial Ambisonic signal mixing and reproduction
- •6.4.1 Principle of spatial Ambisonics
- •6.4.2 Some examples of the first-order spatial Ambisonics
- •6.4.4 Recreating a top virtual source with a horizontal loudspeaker arrangement and Ambisonic signal mixing
- •6.5 Advanced multichannel spatial surround sounds and problems
- •6.5.1 Some advanced multichannel spatial surround sound techniques and systems
- •6.5.2 Object-based spatial sound
- •6.5.3 Some problems related to multichannel spatial surround sound
- •6.6 Summary
- •7.1 Basic considerations on the microphone and signal simulation techniques for multichannel sounds
- •7.2 Microphone techniques for 5.1-channel sound recording
- •7.2.1 Outline of microphone techniques for 5.1-channel sound recording
- •7.2.2 Main microphone techniques for 5.1-channel sound recording
- •7.2.3 Microphone techniques for the recording of three frontal channels
- •7.2.4 Microphone techniques for ambience recording and combination with frontal localization information recording
- •7.2.5 Stereophonic plus center channel recording
- •7.3 Microphone techniques for other multichannel sounds
- •7.3.1 Microphone techniques for other discrete multichannel sounds
- •7.3.2 Microphone techniques for Ambisonic recording
- •7.4 Simulation of localization signals for multichannel sounds
- •7.4.1 Methods of the simulation of directional localization signals
- •7.4.2 Simulation of virtual source distance and extension
- •7.4.3 Simulation of a moving virtual source
- •7.5 Simulation of reflections for stereophonic and multichannel sounds
- •7.5.1 Delay algorithms and discrete reflection simulation
- •7.5.2 IIR filter algorithm of late reverberation
- •7.5.3 FIR, hybrid FIR, and recursive filter algorithms of late reverberation
- •7.5.4 Algorithms of audio signal decorrelation
- •7.5.5 Simulation of room reflections based on physical measurement and calculation
- •7.6 Directional audio coding and multichannel sound signal synthesis
- •7.7 Summary
- •8.1 Matrix surround sound
- •8.1.1 Matrix quadraphone
- •8.1.2 Dolby Surround system
- •8.1.3 Dolby Pro-Logic decoding technique
- •8.1.4 Some developments on matrix surround sound and logic decoding techniques
- •8.2 Downmixing of multichannel sound signals
- •8.3 Upmixing of multichannel sound signals
- •8.3.1 Some considerations in upmixing
- •8.3.2 Simple upmixing methods for front-channel signals
- •8.3.3 Simple methods for Ambient component separation
- •8.3.4 Model and statistical characteristics of two-channel stereophonic signals
- •8.3.5 A scale-signal-based algorithm for upmixing
- •8.3.6 Upmixing algorithm based on principal component analysis
- •8.3.7 Algorithm based on the least mean square error for upmixing
- •8.3.8 Adaptive normalized algorithm based on the least mean square for upmixing
- •8.3.9 Some advanced upmixing algorithms
- •8.4 Summary
- •9.1 Each order approximation of ideal reproduction and Ambisonics
- •9.1.1 Each order approximation of ideal horizontal reproduction
- •9.1.2 Each order approximation of ideal three-dimensional reproduction
- •9.2 General formulation of multichannel sound field reconstruction
- •9.2.1 General formulation of multichannel sound field reconstruction in the spatial domain
- •9.2.2 Formulation of spatial-spectral domain analysis of circular secondary source array
- •9.2.3 Formulation of spatial-spectral domain analysis for a secondary source array on spherical surface
- •9.3 Spatial-spectral domain analysis and driving signals of Ambisonics
- •9.3.1 Reconstructed sound field of horizontal Ambisonics
- •9.3.2 Reconstructed sound field of spatial Ambisonics
- •9.3.3 Mixed-order Ambisonics
- •9.3.4 Near-field compensated higher-order Ambisonics
- •9.3.5 Ambisonic encoding of complex source information
- •9.3.6 Some special applications of spatial-spectral domain analysis of Ambisonics
- •9.4 Some problems related to Ambisonics
- •9.4.1 Secondary source array and stability of Ambisonics
- •9.4.2 Spatial transformation of Ambisonic sound field
- •9.5 Error analysis of Ambisonic-reconstructed sound field
- •9.5.1 Integral error of Ambisonic-reconstructed wavefront
- •9.5.2 Discrete secondary source array and spatial-spectral aliasing error in Ambisonics
- •9.6 Multichannel reconstructed sound field analysis in the spatial domain
- •9.6.1 Basic method for analysis in the spatial domain
- •9.6.2 Minimizing error in reconstructed sound field and summing localization equation
- •9.6.3 Multiple receiver position matching method and its relation to the mode-matching method
- •9.7 Listening room reflection compensation in multichannel sound reproduction
- •9.8 Microphone array for multichannel sound field signal recording
- •9.8.1 Circular microphone array for horizontal Ambisonic recording
- •9.8.2 Spherical microphone array for spatial Ambisonic recording
- •9.8.3 Discussion on microphone array recording
- •9.9 Summary
- •10.1 Basic principle and implementation of wave field synthesis
- •10.1.1 Kirchhoff–Helmholtz boundary integral and WFS
- •10.1.2 Simplification of the types of secondary sources
- •10.1.3 WFS in a horizontal plane with a linear array of secondary sources
- •10.1.4 Finite secondary source array and effect of spatial truncation
- •10.1.5 Discrete secondary source array and spatial aliasing
- •10.1.6 Some issues and related problems on WFS implementation
- •10.2 General theory of WFS
- •10.2.1 Green’s function of Helmholtz equation
- •10.2.2 General theory of three-dimensional WFS
- •10.2.3 General theory of two-dimensional WFS
- •10.2.4 Focused source in WFS
- •10.3 Analysis of WFS in the spatial-spectral domain
- •10.3.1 General formulation and analysis of WFS in the spatial-spectral domain
- •10.3.2 Analysis of the spatial aliasing in WFS
- •10.3.3 Spatial-spectral division method of WFS
- •10.4 Further discussion on sound field reconstruction
- •10.4.1 Comparison among various methods of sound field reconstruction
- •10.4.2 Further analysis of the relationship between acoustical holography and sound field reconstruction
- •10.4.3 Further analysis of the relationship between acoustical holography and Ambisonics
- •10.4.4 Comparison between WFS and Ambisonics
- •10.5 Equalization of WFS under nonideal conditions
- •10.6 Summary
- •11.1 Basic principles of binaural reproduction and virtual auditory display
- •11.1.1 Binaural recording and reproduction
- •11.1.2 Virtual auditory display
- •11.2 Acquisition of HRTFs
- •11.2.1 HRTF measurement
- •11.2.2 HRTF calculation
- •11.2.3 HRTF customization
- •11.3 Basic physical features of HRTFs
- •11.3.1 Time-domain features of far-field HRIRs
- •11.3.2 Frequency domain features of far-field HRTFs
- •11.3.3 Features of near-field HRTFs
- •11.4 HRTF-based filters for binaural synthesis
- •11.5 Spatial interpolation and decomposition of HRTFs
- •11.5.1 Directional interpolation of HRTFs
- •11.5.2 Spatial basis function decomposition and spatial sampling theorem of HRTFs
- •11.5.3 HRTF spatial interpolation and signal mixing for multichannel sound
- •11.5.4 Spectral shape basis function decomposition of HRTFs
- •11.6 Simplification of signal processing for binaural synthesis
- •11.6.1 Virtual loudspeaker-based algorithms
- •11.6.2 Basis function decomposition-based algorithms
- •11.7.1 Principle of headphone equalization
- •11.7.2 Some problems with binaural reproduction and VAD
- •11.8 Binaural reproduction through loudspeakers
- •11.8.1 Basic principle of binaural reproduction through loudspeakers
- •11.8.2 Virtual source distribution in two-front loudspeaker reproduction
- •11.8.3 Head movement and stability of virtual sources in Transaural reproduction
- •11.8.4 Timbre coloration and equalization in transaural reproduction
- •11.9 Virtual reproduction of stereophonic and multichannel surround sound
- •11.9.1 Binaural reproduction of stereophonic and multichannel sound through headphones
- •11.9.2 Stereophonic expansion and enhancement
- •11.9.3 Virtual reproduction of multichannel sound through loudspeakers
- •11.10.1 Binaural room modeling
- •11.10.2 Dynamic virtual auditory environments system
- •11.11 Summary
- •12.1 Physical analysis of binaural pressures in summing virtual source and auditory events
- •12.1.1 Evaluation of binaural pressures and localization cues
- •12.1.2 Method for summing localization analysis
- •12.1.3 Binaural pressure analysis of stereophonic and multichannel sound with amplitude panning
- •12.1.4 Analysis of summing localization with interchannel time difference
- •12.1.5 Analysis of summing localization at the off-central listening position
- •12.1.6 Analysis of interchannel correlation and spatial auditory sensations
- •12.2 Binaural auditory models and analysis of spatial sound reproduction
- •12.2.1 Analysis of lateral localization by using auditory models
- •12.2.2 Analysis of front-back and vertical localization by using a binaural auditory model
- •12.2.3 Binaural loudness models and analysis of the timbre of spatial sound reproduction
- •12.3 Binaural measurement system for assessing spatial sound reproduction
- •12.4 Summary
- •13.1 Analog audio storage and transmission
- •13.1.1 45°/45° Disk recording system
- •13.1.2 Analog magnetic tape audio recorder
- •13.1.3 Analog stereo broadcasting
- •13.2 Basic concepts of digital audio storage and transmission
- •13.3 Quantization noise and shaping
- •13.3.1 Signal-to-quantization noise ratio
- •13.3.2 Quantization noise shaping and 1-Bit DSD coding
- •13.4 Basic principle of digital audio compression and coding
- •13.4.1 Outline of digital audio compression and coding
- •13.4.2 Adaptive differential pulse-code modulation
- •13.4.3 Perceptual audio coding in the time-frequency domain
- •13.4.4 Vector quantization
- •13.4.5 Spatial audio coding
- •13.4.6 Spectral band replication
- •13.4.7 Entropy coding
- •13.4.8 Object-based audio coding
- •13.5 MPEG series of audio coding techniques and standards
- •13.5.1 MPEG-1 audio coding technique
- •13.5.2 MPEG-2 BC audio coding
- •13.5.3 MPEG-2 advanced audio coding
- •13.5.4 MPEG-4 audio coding
- •13.5.5 MPEG parametric coding of multichannel sound and unified speech and audio coding
- •13.5.6 MPEG-H 3D audio
- •13.6 Dolby series of coding techniques
- •13.6.1 Dolby digital coding technique
- •13.6.2 Some advanced Dolby coding techniques
- •13.7 DTS series of coding technique
- •13.8 MLP lossless coding technique
- •13.9 ATRAC technique
- •13.10 Audio video coding standard
- •13.11 Optical disks for audio storage
- •13.11.1 Structure, principle, and classification of optical disks
- •13.11.2 CD family and its audio formats
- •13.11.3 DVD family and its audio formats
- •13.11.4 SACD and its audio formats
- •13.11.5 BD and its audio formats
- •13.12 Digital radio and television broadcasting
- •13.12.1 Outline of digital radio and television broadcasting
- •13.12.2 Eureka-147 digital audio broadcasting
- •13.12.3 Digital radio mondiale
- •13.12.4 In-band on-channel digital audio broadcasting
- •13.12.5 Audio for digital television
- •13.13 Audio storage and transmission by personal computer
- •13.14 Summary
- •14.1 Outline of acoustic conditions and requirements for spatial sound intended for domestic reproduction
- •14.2 Acoustic consideration and design of listening rooms
- •14.3 Arrangement and characteristics of loudspeakers
- •14.3.1 Arrangement of the main loudspeakers in listening rooms
- •14.3.2 Characteristics of the main loudspeakers
- •14.3.3 Bass management and arrangement of subwoofers
- •14.4 Signal and listening level alignment
- •14.5 Standards and guidance for conditions of spatial sound reproduction
- •14.6 Headphones and binaural monitors of spatial sound reproduction
- •14.7 Acoustic conditions for cinema sound reproduction and monitoring
- •14.8 Summary
- •15.1 Outline of psychoacoustic and subjective assessment experiments
- •15.2 Contents and attributes for spatial sound assessment
- •15.3 Auditory comparison and discrimination experiment
- •15.3.1 Paradigms of auditory comparison and discrimination experiment
- •15.3.2 Examples of auditory comparison and discrimination experiment
- •15.4 Subjective assessment of small impairments in spatial sound systems
- •15.5 Subjective assessment of a spatial sound system with intermediate quality
- •15.6 Virtual source localization experiment
- •15.6.1 Basic methods for virtual source localization experiments
- •15.6.2 Preliminary analysis of the results of virtual source localization experiments
- •15.6.3 Some results of virtual source localization experiments
- •15.7 Summary
- •16.1.1 Application to commercial cinema and related problems
- •16.1.2 Applications to domestic reproduction and related problems
- •16.1.3 Applications to automobile audio
- •16.2.1 Applications to virtual reality
- •16.2.2 Applications to communication and information systems
- •16.2.3 Applications to multimedia
- •16.2.4 Applications to mobile and handheld devices
- •16.3 Applications to the scientific experiments of spatial hearing and psychoacoustics
- •16.4 Applications to sound field auralization
- •16.4.1 Auralization in room acoustics
- •16.4.2 Other applications of auralization technique
- •16.5 Applications to clinical medicine
- •16.6 Summary
- •References
- •Index

206 Spatial Sound
Various weights and combinations of errors lead to different optimized decoding coefficients. For example, at low frequencies, the reproduced sound pressure, the velocity localization vector-based azimuth, the velocity vector magnitude, and the energy localization vectorbased azimuth should be optimized preferentially. Accordingly, the combination of Err1, Err3, Err5, and Err6 is preferentially chosen as the cost function. At mid-high frequencies, the reproduced power, energy vector magnitude, velocity localization vector-based azimuth, and energy localization vector-based azimuth should be optimized preferentially. Accordingly, the combination of Err2, Err4, Err5, and Err6 are preferentially chosen as the cost function. This choice of a frequency-dependent cost function leads to two sets of the decoding equations for lowand mid-high frequencies.
In the Tabu search, given the form of the overall cost function, decoding coefficients are initialized at some random values (or some predetermined values that have been derived from other methods). Each coefficient is increased or decreased at a predetermined step size in a given order but is restricted within a predetermined bound. Then, the variation in the overall cost (error) function is evaluated. Decoding coefficients change toward the direction of a decreasing overall cost function, and the best results are kept. A convergent result is obtained by a recursive search. On the basis of the initial coefficients expressed in Equation (5.2.28), Wiggins derived a set of decoding coefficients by using a Tabu search. In comparison with the decoding coefficients given in Equation (5.2.28), decoding with Tabu-search-based coefficients improves the accuracies in a virtual source direction (especially for the front source) and power level, although the maximum rv and rE decrease slightly.
New optimized criteria, such as uniformity and standard deviation in localization, have also been supplemented to the cost function for solving Ambisonic-decoding coefficients (Moore and Wakefield, 2008). Accordingly, the cost function is constructed by supplementing the weighted combination of the standard deviations of Err3, Err4, Err5, and Err6 over the L target directions into Equation (5.2.30). The optimized procedure for the second-order Ambisoniclike signal mixing has been written as software (Heller et al., 2010). Some other mathematical algorithms, such as artificial neuron networks and genetic algorithms, have also been used for nonlinear optimization to derive the Ambisonic-like decoding equations and coefficients of irregular loudspeaker configurations (Tsang and Cheung, 2009; Tsang et al., 2009).
Some studies have indicated that all aforementioned optimized criteria are difficult to satisfy for the second-order Ambisonic-like signal mixing via an ITU5.1-channel loudspeaker configuration. Four loudspeaker configurations are relatively appropriate for the secondorder Ambisonic-like signal mixing. Adding a front-center loudspeaker is not always beneficial to the lateral virtual source for the second-order Ambisonic-like signal mixing. Therefore, region-dependent signal mixing methods can be used. Three front loudspeakers with pairwise amplitude panning (or local Ambisonic-like mixing discussed in Section 5.2.4) can be used to recreate the virtual source within the front region of −30° ≤ θS ≤30°. The four loudspeakers of L, R, LS, and LS with Ambisonic-like signal mixing are used to recreate lateral and rear virtual sources. The design of this signal mixing method is relatively simple. For example, a frequency-dependent gain is designed to reduce the variation in the virtual source position with frequency (Xie, 2001a). Heller et al. (2010) illustrated more examples for four horizontal loudspeakers with irregular configurations.
5.2.4 Optimization of three frontal loudspeaker signals and local Ambisonic-like signal mixing
In multichannel sound reproduction, a horizontal plane can be divided into subregions, and the virtual source in each subregion is recreated by loudspeakers in the same subregion. The loudspeaker signals in each subregion can also be optimized. For local sound field signal

Multichannel horizontal surround sound 207
mixing or local Ambisonic-like signal mixing, the normalized loudspeaker signals in each subregion are a linear combination of azimuthal harmonics. This mixing differs from global Ambisonic (-like) signal mixing. In the latter, the normalized signals of all loudspeakers are a linear combination of azimuthal harmonics.
For 5.1-channel sound, the virtual source in the frontal region is recreated by the three frontal loudspeakers. On the basis of the virtual sound localization theorem, Gerzon (1990) first derived the local Ambisonic-like signal mixing for three frontal loudspeakers. Generally, similar to Equation (5.2.22), Equation (5.2.31) expresses the normalized signals for three frontal loudspeakers with the first-order local Ambisonic-like signal mixing:
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AL S Atotal D0,1L D1,1L |
cos S D1,2L sin S |
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AR S Atotal D0,R D1,R cos S D1,R sin S |
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AC S Atotal D0,C D1,C cos S D1,C sin S |
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Form |
the |
left–right symmetry, decoding |
coefficients satisfy |
D 1 |
D 1 |
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D 1 , |
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D 2 |
D 2 |
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0,L |
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0 . If the optimization at low frequencies is considered only, the criteria |
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of the optimized interaural phase delay difference and its variation with the head rotation or the equivalent criteria of the optimized velocity localization vector in Equation (5.2.31) are used to derive the decoding coefficients. Gerzon (1990) presented the results for three frontal loudspeakers arranged at 0° and ±45°. Here, the results for the frontal loudspeakers with an ITU configuration are given. The decoding coefficients are derived by substituting Equation (5.2.31) into Equation (3.2.22) and letting θv = θS and rv = 1. Then, the normalized loudspeaker signals in Equation (5.2.31) become
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cos S 2 |
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AL Atotal 1 |
3 sin S |
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cos S 2 |
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AR Atotal 1 |
3 sin S |
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AC Atotal |
3 2cos S |
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Similar to Equation (4.3.41), Equation (5.2.33) shows Atotal(overall gain) for constantamplitude normalization:
Atotal |
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(5.2.33) |
Atotal for constant-power normalization is slightly complicated. Equation (5.2.32) proves
that a target-azimuth-independent Atotal results in an overall power of Pow′ = AL2 + AR2 + AC2 that varies with the target azimuth θS. The virtual source direction depends on the relative
magnitude (and phase or time) relationships among the loudspeaker signals. As such, a tar-
get-azimuth-dependent Atotal can be chosen so that the overall power of loudspeaker signals is normalized to a unit, then
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208 Spatial Sound
Figure 5.8 The first-order local Ambisonic-like panning curves for three frontal loudspeakers.
This set of signals is difficult to be created with a practical microphone recording technique, but it can be easily simulated via signal processing.
Figure 5.8 illustrates the first-order local Ambisonic-like panning curves for the three frontal loudspeakers. The signals are normalized according to Equation (5.2.34). For a target source at each loudspeaker direction (θS = 0° or ±30°), the signals for two other loudspeakers vanish. In this case, the virtual source is recreated by a single loudspeaker without a crosstalk (with infinite interchannel separation). This ideal feature is similar to that in pair-wise amplitude panning. However, for the first-order local Ambisonic-like panning, the virtual source between two loudspeakers is recreated by three loudspeakers, and the signal for the third loudspeaker is out of phase. This feature is found in Ambisonic-like panning. As stated in Sections 3.2.2 and 4.1.3, an out-of-phase signal is essential to ensure that the velocity vector magnitude is rv = 1 and to stabilize the virtual source. Equations (3.2.7) and (3.2.9) prove that the perceived virtual source direction matches with that of the target source at low frequencies and within the region of −30° ≤ θ ≤ 30°.
Equation (3.2.34) verifies that the energy localization vector-based azimuth θE is inconsistent with the velocity localization vector-based azimuth θv for signal mixing given by Equation (5.2.32). Gerzon (1990) also pointed out that the mismatch of θE and θv may lead to the directional distortion of a virtual source at high frequencies, such as the movement of a high-frequency virtual source near the front toward the direction of central–frontal loudspeakers. Gerzon (1992c) further proposed an optimized method through which the velocity localization vector at low frequencies and the energy localization vector at mid-high frequencies are considered. That is, the decoding coefficients are chosen so that θE matches with θv.
The aforementioned optimized signal mixing for three frontal loudspeakers is theoretically perfect, but it is rarely used in practical program production.
5.2.5 Time panning for 5.1-channel surround sound
Time panning for 5.1-channel surround sound is similar to the case of a two-channel stereophonic sound. Some studies have investigated the possibility of using pair-wise time panning to recreate a virtual source in 5.1-channel reproduction with transient stimuli. Using female speech as a stimulus, Martin et al. (1999) explored the summing localization with ICTD only in the 5.1-channel loudspeaker configuration. They arranged the three frontal loudspeakers identical to those in Figure 5.2, but they arranged two surround loudspeakers in azimuths of±120°.