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Am Bio Phonics

The document describes the Ambiophonics method for creating a realistic spatial reproduction of music recordings. Ambiophonics uses a Stereo Dipole loudspeaker pair driven by cross-talk cancellation filters to reproduce direct sound, and a surround array convolved with measured room impulse responses to recreate early reflections and reverberation. Subjective tests showed Ambiophonics provided a more realistic listening experience than a Stereo Dipole alone or a system using virtual ambisonic microphones. The method can be implemented cost-effectively using modern digital signal processing and convolution techniques.

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189 views21 pages

Am Bio Phonics

The document describes the Ambiophonics method for creating a realistic spatial reproduction of music recordings. Ambiophonics uses a Stereo Dipole loudspeaker pair driven by cross-talk cancellation filters to reproduce direct sound, and a surround array convolved with measured room impulse responses to recreate early reflections and reverberation. Subjective tests showed Ambiophonics provided a more realistic listening experience than a Stereo Dipole alone or a system using virtual ambisonic microphones. The method can be implemented cost-effectively using modern digital signal processing and convolution techniques.

Uploaded by

Laura Nataly Villa
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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Ambiophonic Principles for the Recording and Reproduction of Surround Sound for Music

Angelo Farina (1), Ralph Glasgal (2), Enrico Armelloni (1), Anders Torger (1) (1) Industrial Engineering Dept., University of Parma, Via delle Scienze 181/A Parma, 43100 ITALY HTTP://pcfarina.eng.unipr.it (2) Ambiophonics Institute, 4 Piermont Road, Rockleigh, New Jersey 07647, USA HTTP://www.ambiophonics.org

The Ambiophonics method


Ambiophonics is an hybrid method for creating a realistic spatial reproduction of staged music, starting from two-channel recordings, but extensible to various kinds of microphonic arrangements up to discrete multichannel The system is based on two indipendently designed groups of loudspeakers: a Stereo Dipole, responsible for the reproduction of the direct sound and early reflections coming from the stage, asnd a surround periphonic array, driven by realtime convolution with room impulse responses

The Stereo Dipole


Original 2-channels recording of the signals coming from N sources
dNr N 3 2 d2l 1 d1r d1l dNl d2r xl xr

CD recording

10

Cross-talk canceller

Reproduction over the Stereo-Dipole loudspeaker pair

The cross-talk cancellation allows for the replica of the recorded signals at the ears of the listener

The surround convolution


Measurement of two independent sets of 3D impulse responses MLS signal sR
R

3D microphone probe

sL,3D
L sL

sR,3D

Original stereo recording


z 3 4 2 1 r y 6 7 8 5

Multichannel convolver

Multichannel convolver

Reproduction over the Ambisonics loudspeaker array

The cross-talk cancellation allows for the replica of the recorded signals at the ears of the listener

Design of cross-talk canceling filters First, a binaural measurement is made in front of the Stereo Dipole loudspeakers
convolver

xl
Binaural stereo signal

fll flr frl yl frr yl


yr

xr

yr R hlr hrl hrr pl pr

hll

Then, the cross-talk cancelling filters are computed, so that their convolution with the measured impulse responses reduces to the identity matrix

Theory of cross-talk canceling filters


f ll = (hrr ) InvDen f = ( h ) InvDen lr lr f rl = ( hrl ) InvDen f = (h ) InvDen ll rr InvDen = InvFilter (hll hrr hlr hrl ) C ( ) = FFT (hll ) FFT (hrr ) FFT (hlr ) FFT (hrl )

Conj[C ( )] InvDen( ) = Conj[C ( )] C ( ) + ( )

The regularization parameter, , has to be adjusted by trials

Example

Measured impulse responses h

Plugin for CoolEdit which computes the inverse filters f

Example

Computed long-FIR inverse filters f

Verification of the cross-talk cancellation

Warped FIR cross-talk cancellation


Todays DSP boards are not powerful enough for convolving long inverse FIR filters Warping can be used for concentrating the computing power in the frequency range where it is most needed
z = A ( ) =

+ 1+

=0
= 0.0

As the processing network is more involved than a traditional FIR, the number of taps which can be used is lower. Nevertheless, the perceived performances, on a given DSP board, are usually better than with a normal FIR

Warped FIR implementation on a SHARC


The WFIR structure was coded in assembly on the AD21061 and on the AD21065L processors: here the assembly code of the main cycle is shown:
LCNTR=Wfilter_taps-1 , DO wmac_rr UNTIL LCE; F12=F2*F4, F9=dm(I5,M7), F4=pm(I9,M8); F10=F2*F5, F8=F8+F12, F9=dm(I5,M6); F1=F9-F10, F9=dm(I5,0); F10=F1*F7, dm(I5,M7)=F2; wmac_rr:F2=F9+F10;

It takes 5 lines (CPU cycles), whilst the normal FIR is just one line thus a SHARC can run a set of 4 FIRs of more than 200 taps each, but only 42 taps with WFIR.

Subjective blind comparison: FIR vs. WFIR


14 normal-hearing subjects (6 females, 8 males) Two sound samples: binaural recording of natural sounds and a piece of pop music (Elton John) 5-levels scale (insufficient, mediocre, sufficient, fair, good ) The listener was free to switch at will between the two processing algorithms, denoted simply as A and B Classic ANOVA analysis of the subjective response
Question Avg. A Avg. B Anova's F Prob. factor 4.79 4.36 4.71 4.57 4.36 4.07 4.43 34.47 4.38 21.72 10.88 11.56 1.60 0.98 0.00% 4.63%
Score

Averages, standard deviations and ANOVA probability results


6

5.5

0.00%

4.63%

0.01%

0.28%

0.22%

21.71%

33.10%

Overall appreciation 3.57 Image localization Stage width Naturality 3.79 3.50 3.71

4.5

0.01% 0.28% 0.22% 21.7% 33.1%

3.5

Low frequency resp. 3.29 Mid frequency resp. 3.79 Hi frequency resp. 4.14

2.5

1-Overall appreciation

2-Image localization

3-Stage amplitude

4-Naturality
Question A - FIR

5-Low frequency 6-Mid frequency 7-High resp. resp. frequency resp.

B - WFIR

Virtual Ambisonics surround


Measurement of 3D (B-format) impulse responses in theatres, with two source positions on the stage The IRs are processed, deriving the responses of several directive microphones Each soundtrack of the original stereo recording is convolved with the corresponding IR For each loudspeaker, the results of the two convolutions are mixed

Z Px-

X Px+ d

7 spaced omni

Soundfield MK-V

B&K WA0447

Measurements in 3 Italian theatres

La Scala, Milan
2.5

T. Comunale, Ferrara T. Verdi, Trieste


Reverberation Times

1.5 T20 (s) 1 0.5 0 63 125 250 500 1000 Frequency (Hz) 2000 4000 8000 16000

Comunale Ferrara

La Scala Milano

Verdi Trieste

Synthesis of directive microphones


The WXYZ channels of a B-format IR can be processed, extracting a single (mono) response of a virtual microphone pointing along a given versor r (rx, ry, rz):
r 1 V (r ) = (2 D ) W + D (rx X + ry Y + rz Z ) 2

The directivity factor D can assume the following values: D=0.0 omnidirectional D=0.5 subcardioid D=1.0 cardioid + + = D=1.5 hypercardioid D=2.0 figure-of-eight + =

The Double-reverberation problem


When an impulse response is reproduced in another reverberant space, the resulting reverberant tail is the convolution of the two reverberant tails
The problem can be compensated for, by manual editing the first impulse response, applying (with CoolEdit) a time.varying amplitude shaping:
Ts = 0.5 s
+

Ts = 1.0 s

Ts = 1.13 s

Hardware implementation
A complete Ambiophonics system can be implemented, nowadays, coupling a general-purpose DSP unit (cross-talk cancellation) and convolution-based reverberators
Professional CD Player

Stereo Dipole

BSS SoundWeb 9088-ii

Here the cross-talk cancellation network implemented on the Soundweb is shown:


+

2 x Sony DRES-777

8 Surround speakers

The limit of this system is in the number of coefficients of both the cross-talk canceler and the room convolver

Software implementation
The preferred implementation is by means of a simple software convolver and a cheap, modern PC. Two solutions are currently available:

Ambiovolver is a program running on any Windows PC equipped with a multichannel soundboard

BruteFIR is an highly optimized Linux program running on a customized, noiseless computer with a digital RME audio interface

Both systems are being demonstrated in room 22

Latency vs. performance


The software implementation is based on frequency-domain convolution (overlap-and-save), which inherently introduces some latency. Furthermore, the audio stream I/O on a PC is always buffered, so an intrinisic latency is caused by the buffer size BruteFIR distinguishes himself from other convolvers by the fact that it implements partitioned convolution: the impulse response is subdivided in many segments of equal length, and this reduces the latency to twice the length of a segment, instead of twice the length of the whole IR. On modern CPUs, the partitioned convolution is more efficient than traditional unpartitioned overlap-and-save, with a reduction of CPU load of 20-50%, and can reduce the overall latency to less than 100 ms. Very efficient FFT implementations are freely available (Intel NSP, FTTW), and thus the computing power of a PC is enough for real-time convolution of 20 IRs, at 44.1 KHz, 32 bits, each being 65,536 points long. The demonstration machine, installed in room 22, is an old Pentium-II 400 MHz.

Subjective comparative experiment


9 normal-hearing subjects (males) Three sound samples: Simple ranking test between three systems: Stereo-Dipole, Virtual Ambisonics, complete Ambiophonics Each listener can switch freely among the three systems during the playback

Music piece Mozart, Te Deum K141, Sennheiser MKE2002 (Mozart Sacro, n. 1) Buxtehude KFM -6 (Ambiopole demo 1, n.13) Mozart, Overture Le nozze di Figaro, bars 1 -50, ORTF (Denon PG 6006, n. 37)

Theatre La Scala

Cross-talk filters Binaural

Teatro Comunale Teatro Verdi

Sphere ORTF

Results
Method Avg. Score Stereo Dipole 1.99 Virtual Ambiophonics Ambisonics 2.77 1.24

Conclusions
Ambiophonics revealed to give significant advantages over the two surround systems which constitutes it. It recreates a realistic virtual acoustic space by means of convolution with proper digital filters The computational power required can be obtained cheaply by means of a modern PC The system can be configured for different number and position of the loudspeakers The sweet spot can easily accomodate three persons, and also far from this area, the overall acoustic impression remains that of being in a concert hall.

Internet Links
- The CoolEdit plugins, employed for measuring and processing the impulse responses, can be downloaded freely from HTTP://www.ramsete.com/aurora - The sets of impulse responses, and the sound samples employed for the subjective tests are available for download at HTTP://pcangelo.eng.unipr.it/public/AES19 - The programs for computing a B-format IR from 7 spaced omnis, and for deriving the virtual directive microphones, can be freely downloaded from HTTP://pcangelo.eng.unipr.it/public/B-format/software - The BruteFIR convolver for Linux can be freely downloaded, with its source code, from HTTP://www.ludd.luth.se/~torger/brutefir.html

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