
- •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
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Two-channel stereophonic sound 111 |
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Heq f |
1 |
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c |
. |
(2.2.57) |
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jklm |
j2 f lm |
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In addition, Blumlein originally used an analog circuit with the following response to equalize AS
Heq f 1 |
1 |
, |
(2.2.58) |
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j2 f m |
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where τm is the adjustable parameter. At low frequencies, this equalization and the inverse MS transformation also yielded stereophonic signals with ICLD only.
2.2.5 Spot microphone and pan-pot technique
Multiple spot microphones are used to capture source signals, and each microphone separately captures the signal of each source or each set of sources, resulting in multiple monosource signals. As illustrated in Figure 2.18(a), the mono signal from each spot microphone is split into two channel signals by a pan-pot. A conventional pan-pot is a dual-ganged variable resistor that controls the relative magnitude or level of the mono signal fed to the left and right channels and then leads to a different ICLD. In this case, the desired directional information of a target source is simulated or synthesized artificially. Currently, the equivalent function of a pan-pot can be easily implemented via digital signal processing.
Usually, the overall power of two channel signals is normalized to a constant (unit) to ensure equal-loudness virtual sources in different directions. Therefore, the normalized amplitudes of two signals from the pan-pot are given by
AL sin |
AR cos |
AL2 AR2 1, |
(2.2.59) |
where 0° ≤ ξ ≤ 90° is a parameter. Figure 2.18(b) illustrates the variation in AL and AR with ξ. From Equation (2.1.6), for a head fixed to the frontal orientation, the direction of a low-
frequency virtual source is related to ξ as follows:
sin I |
tan 1 |
sin 0. |
(2.2.60) |
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tan 1 |
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Similarly, according to Equation (2.1.10), for a head oriented to the virtual source, the direction of virtual source is evaluated by
tan I |
tan 1 |
tan 0. |
(2.2.61) |
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tan 1 |
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In both cases, when ξ changes continuously from 0° to 90°, signal amplitude AR decreases continuously from 1 to 0, and AL increases continuously from 0 to 1. Accordingly, the virtual source moves continuously from −θ0 (the direction of the right loudspeaker) to θ0 (the direction of the left loudspeaker). It has AL = AR = 0.707 for ξ = 45°, i.e., a −3 dB drop off compared with the maximal amplitude of a unit value. In this case, the virtual source lies in the directly front direction of θI = 0°.

112 Spatial Sound
Figure 2.19 Two-channel stereophonic panning curves.
For constant-power panning, the normalized amplitudes of two channel signals can be derived from Equations (2.2.60) or (2.2.61) subjected to the condition of a constant (unit) overall power. For a head fixed to the front orientation, we have
AL |
2 |
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sin 0 sin I |
AR |
2 |
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sin 0 sin I |
. |
(2.2.62) |
2 |
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sin2 0 sin2 I |
2 |
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sin2 0 sin2 I |
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For a head rotating to the orientation of the virtual source, we have
AL |
2 tan 0 tan I |
AR |
2 tan 0 tan I |
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. |
(2.2.63) |
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2 |
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tan2 0 |
tan2 I |
2 |
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tan2 0 |
tan2 |
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I |
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Equation (2.2.62) or Equation (2.2.63) gives normalized amplitudes of two channel signals as functions of a virtual source direction, i.e., signal panning function. Figure 2.19 plots the panning functions, i.e., the panning curve of the left and right channel signals for a stereophonic loudspeaker configuration with a span angle of 2θ0 = 60°. In the front and either loudspeaker directions (0°, ±30°), Equations (2.2.62) and (2.2.63) yield identical results. The two equations yield different results in other directions. However, the difference is trivial if the span angle 2θ0 between two loudspeakers does not exceed 60°.
Applying the transformation ξ = θS+ 45° with −45° ≤ θS ≤ 45°, Equation (2.2.59) becomes
AL cos S 45 |
AR cos S 45 . |
(2.2.64) |
This expression is consistent with Equation (2.2.11). Therefore, the nature of synthesizing two-channel stereophonic signals with pan-pot is equivalent to the artificial simulation of the signals from a coincident bidirectional XY microphone pair for a source within the effective recording range. Here, θS is the azimuth of a target source in the original sound field to be simulated rather than the direction of the perceived virtual source in reproduction shown in Equation (2.2.12).

Two-channel stereophonic sound 113
In addition to constant-power panning, two-channel signals are sometimes normalized according to the condition of constant unit amplitude, i.e., constant-amplitude panning:
AL AR 1. |
(2.2.65) |
The recorded signals in Equation (2.2.37) satisfy the condition of constant amplitude. The constant-amplitude and constant-power panning are relatively appropriate for reproduction in anechoic rooms and rooms with some reverberation, respectively.
In practice, the acoustic characteristics of reproduction rooms are usually frequency dependent. Some studies have introduced a frequency-dependent normalization for two channel signals according to the acoustic characteristics of a reproduction room (Laitinen et al., 2014), i.e.,
AL AR 1, |
(2.2.66) |
where 1 ≤ λ = λ(f, DTT) ≤ 2 is a parameter depending on frequency (band) and direct-to-total energy ratio (DTT). The DTT can be evaluated using the method in Section 1.2.4. λ =1 and
λ=2 corresponds to constant amplitude and constant-power panning, respectively. Equations (2.2.62) and (2.2.63) are derived from the stereophonic laws of sine and tan-
gent respectively. For practical music stimuli, however, the direction of the perceived virtual source may not exactly match the results of the laws of sine and tangent. Therefore, Lee and Rumsey (2013) derived the relationship between the direction of the perceived virtual source and ICLD based on the fitting of the results of a localization experiment for music stimuli and used this relationship for panning curve design.
2.2.6 Discussion on microphone and signal simulation techniques for two-channel stereophonic sound
Various microphone and signal simulation techniques for two-channel stereophonic sound are presented in the previous sections. These techniques are chosen and used flexibly according to practical requirements.
The MS, XY, and near-coincident microphone techniques are usually chosen for large orchestra recording to achieve a fused sensation in reproduction. The microphone technique and associated parameters, such as the type, directivities, distance to source (orchestra), effective recording angle of coincident microphone pairs, or various parameters of a near-coin- cident microphone pair, are chosen on the basis of practical conditions. The directivities for some coincident microphone products are adjustable and therefore convenient for practical uses.
The performance of XY and MS microphone pairs is compared in some studies (Hibbing, 1989). Although XY and MS microphone pairs are theoretically equivalent, the MS microphone pair is relatively flexible in practical use. Deriving various XY-equivalent signals from a pair of MS signals is relatively easy, freeing from the restriction on available XY microphone products with the desired directivity. At the same time, a practical directional microphone usually possesses the perfect or desired directivity below a certain frequency. As frequency increases, the main lobe of microphones usually narrows. As a result, the high-frequency output magnitude of the directional microphone decreases for an off-axis source, giving rise to timbre coloration in reproduction. This phenomenon occurs in the direct front source in XY recording, especially in an XY recording with a wide span angle between the main axis orientations of two microphones, because the source lies at the off-axis direction of two

114 Spatial Sound
microphones. This problem can be avoided in MS recording because the main axis of the M microphone always points in the front direction. Indeed, the effectiveness of the MS pair in reducing timbre coloration depends on the relative importance of the timbre of the front source at the overall stereophonic stage.
From the preceding analysis on XY coincident microphone pairs (or equivalent MS microphone pairs) and near-coincident microphone pairs, the span angle between the main axes of two microphones, effective recording range, and the span angle between two loudspeakers in reproduction are usually not coincident. The span angle between the main axes of two microphones is just a parameter related to microphone configuration. For sound sources within the effective recording range, the virtual sources in reproduction are limited or mapped to a range between two loudspeakers, when the case of outside-boundary virtual source is not considered. For many living recording stereophonic program materials, virtual source positions in reproduction are not exactly consistent with those of the actual source (such as instruments) at the original stage. However, this consistency is not vital because listeners usually do not care about the absolute positions of sound (virtual) sources in reproduction. Recreating the relative position distribution of virtual sources in reproduction is enough.
For on-site stereophonic recording, one important step is to choose the effective recording range. The effective recording range is determined by the width of source stage (span angle of source distribution with respect to microphones). A wide source stage requires a wide effective recording range. In some experiences from on-site recording, for a narrow sound stage (such as quartet), the effective recording range is chosen to be about 10% wider than the total sound stage to leave a side room (Williams and Du, 2001). However, for a wide sound stage (such as an orchestra), the effective recording range is chosen to be about 10% smaller than the sound stage to enable better resolution of the central orchestra. For an excessively wide sound stage, microphones can be placed backward at a more distant position from the sources to reduce the span angle of source distribution with respect to microphones so that all sources at the sound stage can be recorded by a pair of microphones with a smaller effective recording range. However, when the microphone pair is placed at a more distant position from the sources, the captured power of reverberation sound increases compared with that of the direct sound. In this case, a microphone pair with appropriate directivities, such as a cardioid pair in Figure 2.8(a), can be used to reduce the relative proportion of reverberation components in captured signals. Therefore, the choice of the effective recording range, the distance between a source and microphones, and the directivities and orientation of the main axes of microphones are closely related and restrained. An appropriate choice should be on the basis of practical conditions.
In some situations, stereophonic recording with a coincident or near-coincident microphone pair may not satisfy the requirement from the point of acoustics. In these cases, some additional microphones (or microphone sets) are needed. As the first example, for some music program recordings, such as solo in a concerto, the solo should be enhanced from the background of orchestral music. In this case, in addition to a coincident or near-coincident microphone pair for recording orchestral music, an individual microphone is used to capture the source signal of a solo, and the resultant signal is mixed to the two channel signals by pan-pot (for large solo instruments, such as a piano, a pair of additional microphones are needed). As a second example, for an orchestra with excessive width, instruments at two sides of the orchestra are located at a more distant distance from the coincident microphone pair and lead to a low magnitude in microphone outputs. In this case, a pair of outrigger microphones can be supplemented to the two sides to capture the source signals from the instruments at two sides of the orchestra and the outputs of an outrigger microphone pair are mixed with those of the main microphone pairs. The virtual source of the instruments at the two sides lies in the position of two loudspeakers in reproduction because of the