Apple Computer Inc. v. Burst.com, Inc.

Filing 109

Declaration of Sheila Hemami in Support of 107 Response Burst.com, Inc's Opposition to Plaintiff Apple Computer, Inc.'s Motion for Summary Judgment on Invalidity Based on Kramer and Kepley Patents filed byBurst.com, Inc.. (Attachments: # 1 Exhibit A to S. Hemami Declaration# 2 Exhibit B to S. Hemami Declaration# 3 Exhibit C to S. Hemami Declaration# 4 Exhibit D to S. Hemami Declaration# 5 Exhibit E to S. Hemami Declaration# 6 Exhibit F to S. Hemami Declaration# 7 Exhibit G to S. Hemami Declaration# 8 Exhibit H to S. Hemami Declaration# 9 Exhibit I to S. Hemami Declaration# 10 Exhibit J to S. Hemami Declaration# 11 Exhibit K to S. Hemami Declaration)(Related document(s) 107 ) (Crosby, Ian) (Filed on 6/7/2007)

Apple Computer Inc. v. Burst.com, Inc.

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Introduction
he rapid increase i n digital connectivity of telephone networks, brought about by the gradual f removal o most analog links, suggests a new look at enhancing the quality of audio transmitted over the telephone network. Pulse-code modulation (PCM) with 64 kbit/s p-law or A-law (G.711) arose in response to the need for multiple analog-digital-analog conversions of standard 300-3,400-Hzaudio signals. Such signals are considered here to be narrowband audio. Modern speech coding techniques permit reduction of the transmitted bit rate, while preserving audio quality, as in C C I T T (International Telegraph and Telephone Consultative Committee) Recommendation G.721) where the customary 300-3,400-Hz-wide telephone signal is encoded at 32 kbit/s [l]. Alternatively, one can provide improved audio quality a n d maintain the transmission rate at 63 kbitls. Such improvements are most important for audio or audio-visual conferencing applications where one would like to approach the quality of face-to-face communication. C C I T T Study G r o u p XVIII recognized the need for a new international coding standard on high-quality audio to allow interconnection of diverse switching, transmission, and terminal equipment and, thus, organiLed a n Expert G r o u p in 1983 to recommend a n appropriate coding technique. Hummel [2] provides a good introduction to the working methods of the CCITT. T h e coding method described in this paper constitutes the group's recommendation, which was approved by the C C I T T through a n accelerated procedure in 1986. T h e algorithm represents the results of a joint effort of contributors from around the world,* and is best described i n a series of papers presented a t Glohecom '86 [3]. T h i s paper is meant to be a tutorial discussion, responsibility for which lies completely with f the author. Bit-level particulars o the algorithm, although important for correct implementation of the standard, are not discussed i n detail. For more complete information, the reader should refer to the forthcoming C C I T T document.

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G,722, A New CCITT Coding Standard for Digital Transmission of Wideband Audio Signals
Paul Mermelstein

Requirements
T h e main objective for the new standard is to allow speech transmission at 64 kbit/s with quality as high as poissible and significantly better than that provided by 8bits/sample, 8-kHz sampled PCM coding. If the signal is sampled a t 16 kHz, or twice the PCM rate, the spectrum of the signal to be encoded can be extended to about 7 kHz (3-dB point), and this results in a major improve-

*Acomplete list of the participants in the Expert Groupcan
be found in Appendix 1 t o the Report CCITT, COM XVIII-R 17-E, April 1986. Participating organizations included B N K ,

Canada; CNET, France; FTZ, Federal Republic of Germany; CSELT and SIP, Italy; N T T , Fujitsu and KDD, Japan; P ' I I ' , Switzerland; B T , United Kingdom; Bellcore and Comsat, lrnited States. Technical contributions of all participant5 are recognized and acknowledged without specific credit on individual items.
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conlerence bridges is best carried out with a uniformly quantized representation of the digital signal. To allow for multiple bridges in one connection, provision is made for a small number (up to three) of digital encoding/decoding sequences. Furthermore, both narrowband and wideband audio signals may arrive at audio bridges and bridge output should also be available in narrowband or wideband form.

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Auxiliary data

allocating the bits across the different bands, the error variance in the reconstructed signal can he shaped n i t h frequency. With the audio signal subdivided into two I-kHz-wide hands, a high signal-to-noise ratio in the lower band hecomes perceptually more important than in the higher band. An advantage ot ;I design that uses tlvo equally wide subhands is that each compotient is suhsampletl to 8 kHz and the total transmission rate may he reduced in 8-kbit's steps by reducing the number of bits assigned to samples of one or the other hand. While the hit rate may also he reduced by reducing the sampling rate, those processes generally arc more complex to implement. These considerations led to design and e\xluation of tivo alternative subband ADPCM systems, one using 5 and 3 hits'sample for the lolv- and high-band components, respectively, the other, 6 and 2 bits/sample. T h e G.721 ADPCM design employs a n adapti1.r predictor with two poles and six zei-os. A fixed predictor design w i s also tested for the \videhand coder, h i i t i t led to a generally lower speech quality. A time-wrying adaptive allocation of bits to the t i v o su1)hands according to the short-time signal characteristics w a s also tried. For \,oiced sounds carrying significant low-frequency energy, one can assign additional bits to the low h n t l ; for fricative sounds, the atlditional hit m a y be assigned more advantageously to the high band. IIowever, for the twoband, 4-kHz-wide subband dcsign, thr. advantage of a n adaptive bit assignment is only apparent at the .4 low2 high hits samplr. assignment a n d is found too small to \va rra n t the add i t i on a 1 r o m p 1exi t y . Overall block diagrams for thr \videhand encoder and decoder ;ire shoivn in Fig. 2. These blocks are discussed in greater detail in the following sections.

filters ( Q M F ) are used to di\.itle the ~videhantlsigii;rl sampled ;it a 16-kHz rate into t'ivo 8-kH7 s m i p l ~ d components to tic transmitted, ;I low hand and ;I high hand, and reconstruct the ivitlehand signal from its rrceived low- and h i g h - h m d componmts. QMF liltet-s are finite-impulse response, impose ;I fixed tlcla)~ thollr \vi phase distort ion, ;I nd c t i s lire t h a t :i1i x i n g prod I I c t s resulting from subsampling the input signal at the transmitter are canceled at the recei\w. Ho\ve\er, quantization noise components introdric-r.d in coding the low- and high-band signals ma)- not be elirnitiatctl complctely by thr. receiver QMF fi1tr.r. Bccausc the 1c1~c.l f o the high-band component of the signal may be :is m u c h a s 40 dR 1o1vr.r than the low-band componenr, aliasing noise introduced into the high-hand frcqiivnc ies due t o coding the low-hand signal might h e i n a d e q u a t c l ~ ~ masked hy the high-band signal componcwt. ' I o achicw
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Subband Filtering
T h e nominal 3-dB hand of the codcc w a s chosen as 50-7,000 Hz. Two sets of identical quadrature mirror
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:I stop-hand rejection of 60 d B , we employ ;I 24-tap filteidesign, introducing a total signal delay of only 3 tns (sec Fig. 3 ) . T h e resulting signal distortion is helow 1 dB o \ w the 100-6,"M-Hz band. T h e numerical precision with which the partial s u m s in the QMF filters ;ire accumulated has a n important of hearing on the ac~ciiracy the loiv- antl high-band signal components grnerated. T h e overall goal for the \videband signal representa tion (after analog-to-digital conversion a t the input and Iiefore digital-to-analog convrrsion at the o u t p u t ) is a precision of 14 tiits. To this end, the internal coding computations are performed with 16 hits. For the suhhand signals to have 16 significant hits, the partial sum computations were found t o require 21 pi-ecision of 24 bits. Since the widehand sign ;I 1 I S acciirate to 14 hits, the s u m and difference signals to lvhich the QMF filter coefficients are applied arc only precise to 13 bits. To prex'ent the intt-otluction of noise due t o differently specified analysis and synthesis filters, the QMF filter coefficients are also represented with 13 bits.
'

ADPCM Coders
T w o A D P C M coders are required, one for the lowhand signal and one for the high-hand signal. Thecoders employ identical adaptation strategies to modify the
lndkallm

qiiantirers and predictors based on the previously ohser\wl characteristics of the input signal. T h e low- antl high-hand coders are very similar, except for small differences due t o the need to vary the number of hits output h y the loiv-hand coder and the fact that the highhand quantizer output is always 2 hits/sample. 'I'hc adaptive predictor design is bonowed directly frotn that investigated in detail in developing the G.721 standard. T h e two-pole, six-zero design combines good prediction gain for speech with relatively simple stability control. Robust adaptation is assured by leaky integrators allowing the effects of transmission errors to dissipate rapidly [8]. Transmission errors may introduce d i f ferences hetlveen the predictor memories at the transmitter and rvceivcr. Adapting the predictors and q u a n tizers using the residual signal alone and not the reconstrurted signal eiisures that the predictors a t the transmi ttei- and receiver recover tracking rapidly f o i - all signals [9]. T h e adaptive qiiantirer design is also horroived directly from the G.721 standard since the 1 0 ~ ' hand signal cotnponen t resembles the narrowhand speech signal in most of its properties. G.721 employs ;I dual-mode quantizer, a locked or s l o w ~ l y adapting mode for voicehand data signals, and a n unlocked or rapidly adapting mode for speech signals. Since the G.722 standard was not required to encode voice-hand data

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Detailed Block Diagram of the Subband-ADPCM Encoder and Decoder.
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signals, only a single rapidly adapting mode had to be implemented. Thedynamic range of the lo\\-band signal quantizer was set to he the same as for G.721, namely 54 dB. A higher dynamic range is a l l o ~ v e d the higIi-l)atid for quantizer, 66 dB, mostly to accommodate music signals. Robust adaptation is emplo)-ed also for the quantiret, scale factor to combat the effects of transmission crrot~s. Embedded quantirers alloiv for possible stripping of the less significant hits from the quantized signal during transmission b y not making use of those bits in the quantizer adaptation process [IO]. T h e l o w h a n d q u a n tizer anticipates that the one o r two least significant hits m a y he stripped from the transmitted code ~ ' o r d and adapts the quantizer and predictor using only the four most significant bits. This results in 4- and 5-hit quanti~ers that are slightly suboptimal in quantiration noise-to-signal ratio compared to the c~u~intirers that may be designed without this cons tr;t i t i t. T h e cmhedding property requires that the 4- m d 5-hit quantization hoiindaries coincide with a subset of those employed for

t h c 6-hi t q 1i;t ri t i rer. Ma t i y t r;t 11smission systems rcq ui t-e 21 minimal number of zero-one alternatives to maintain s) tic.hroniz~ttioti.?To prevrnt the all-zero code from appeiiritig even in the 4-hit data representation, only 15 quantirer levels are used in that mode; this also restricts t h y higher niodcs to 30 m d 60 levels. Esperinicntal f e\xluations ha\.c shoivti only a fraction o a decibel is lost in quantizing speech signals with the embedded q u a n tirer comp:ired to a n unconstrained quantirer design. In tcrms of subjective performance, the embedded design was not found to be significantly different from the noneinbedded design, e t m aftei- four transcodings. A significant systems a d \ m t a g e resulting from the embedded coder design is that the en( oder i n a y opcrate without regard to the momentary data transmission rcquirements. T h e speech codingand data multiplexing operations are separated logically and possihly even physically. `Thus, data may he introduced ;it a point downstream in the transmission path r e m o \ d froin the encoding terminal. T h e receiver o r decoding terminal

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must, of course, be aware of the amount of data introduced in place of speech information so that it may interpret the respective bits appropriately. A more detailed diagram of the subband-ADPCM encoder and decoder is given in Fig. 4. As illustrated, identical encoding by the transmitter and receiver in each of the three modes of low-band quantization is ensured by stripping the two least significant bits in the feedback paths of the predictor and quantizer. The decoder interprets the received data in accordance with the current mode indication. T h e 6-, 5-, or 4-bit words are converted using 60-, 30-, or 15-level inverse quantizer tables to arrive at the appropriate signal estimate. T h e two-pole, six-zero adaptive predictor data-flow structure is illustrated in greater detail in Fig. 5. Alternative implementations are, of course, available; the figure shows a flowchart for perhaps the most simple configuration.

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Fzg. 7. Average Subjective Quality Ratings for Multiplicative Noise Reference Conditions.

Subjective Performance
Before selecting the final design, the Expert G r o u p carried out a series of subjective experiments with different speech signals (diverse languages), music, differingtransmission conditions, and various hardware coding devices. Starting with four different algorithms, as implemented in hardware, the list was pruned to two, and, finally, one compromise design. T h e discussion o n performance that follows pertains only to the final design as embodied in the recommendation. T h e subjective measure of audio quality adopted is the mean opinion score (MOS) on a five-point scale: excellent, good, fair, poor, bad. Since in the anticipated applications of audio conferencing and. hands-free telephony listening over speakers a n d not handsets is most likely, audio was provided over loudspeakers at a level of 70-dB S P L (sound pressure level) at the listener. Seven different language tapes were processed by the same codec hardware. Listening experiments were

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48

conducted in seven laboratories; the results quoted are the average results obtained. Audio quality is best at 64 kbitls, drops slightly when the transmission rate is reduced to 56 kbit/s, and more significantly with a further reduction to 48 khitls (Fig. 6). However, even at 48 kbit/s, the audio quality is significantly better than narrowband PCM. Received audio quality is only slightly affected by transmission but bit error rates up to lom4, a significant quality drop is noted when the error rate reaches T h e subjective MOS may show differences in speech quality due to speaker and listener effects as well as the language used. T o allow different experimental conditions to be compared more precisely, a set of reference conditions of speech mixed with multiplicative white noise was evaluated by each g r o u p of listeners evaluating the coded speech. In each case, Q w indicates the signalto-noise ratio in decibels. Figure 7 gives MOS scores as a function of Q w . T h e hest mean MOS score of 3.3, obtained under error-free 64-khitIs transmission, corresponds to a reference condition of Q w = 45 dB. Note that the direct uncoded speech is assigned the same MOS rating, implying n o measurable quality degradation due to one stage of wideband coding. In contrast, narrowband PCM (G.712) results in a Q w of 32 dB. T h u s , the overall subjective gain for G.722 coded speech relative to G.712 coded speech is equivalent to some 13 dB of noise reduction. Wideband signals coded at 128 kbitls, 8 hitslsample at a sampling rate of 16 kHL, are found to correspond in quality to a Q w of 38 dB. T h i s finding suggests that roughly 6 dB of quality improvement results from expanding the signal bandwidth to the range of 50-7,000 Hz, and 7 dB of additional improvement is obtained by more precise encoding o that signal. f As the available transmission rate is reduced, the quality becomes slightly degraded. At 56 kbitls, we observe a Q w of 43 dB, a very minimal degradation. At 48 kbitls, Q w is 38 dB, which is still significantly better than the quality of G.712. Whenever data transmission is intermittent, conference participants may not even be aware of the audio-quality variations due to dynamic mode switching.

Mode Initialization and Mode Switching
Although the procedures for mode initialization a n d mode switching will be incorporated into a separate

Fig. 6. Average Subjective Quality Ratings of the Final Algorithm.

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recommendation, they are discussed here to indicate holv the G.722 coding procedures may he applied in practical communication systems. To avoid the need for multiple audio terminals o n one's desk-a high-quality widehand terminal to communicate with other wideband terminals and a normal telephone to reach parties equipped only with narrowhand telephones-most wideband terminals will incorporate a narrowband PCM communication mode. Calls can then be set up with the terminal in the narrowband mode (Mode 0). As soon as the called party answers, a n exchange o flags takes plare. T h e calling terminal f transmits one of two flags denoting its capabilities. Terminals may he of Type 1 , which has only ii 64-khit s t r;i n sm i ssi o n cii pa hi 1 i t y , narrow ha 1 d or wideha n d, or 1 Type 2, which implements a t least 64- a n c l 56-khit s ~videhantlmotles (Modes 1 and 2 ) and possihly even 48 khit ' s (Mode 3 ) . T h e flag sent identifies the terminal type; the calling terminal awaits a similar response flag. D u m b terminals having only ;I narro\vband capability cannot respond to the received flag, which leads the c;illing terminal t o conclude, after a suitable time-out, that it must remain in the narrowband mode. Intelligent term i na 1s ;icknoivlcdge t lie received flag by t ra 11smi t t i rig the appropriate response flag and adopt ;I widehand inotle (Mode 1). On receipt of that flag, the calling terminal assumes the same wideband mode. If the exchange of flags takes place on the least significant bit or hit 8 o f every word, it introduces only ;I slight a m o u n t of noise into the audio path. T h u s , narrowband communication is availa tile even during the flag exchange. T h i s plan allows widehand terminals to he introduced gradually into the telephone network, retaini n g connectivity and gradurllly enhancing quality with increased penetration of ividehand terminals. Many North American network connections optionally modify the least significant bit of PCM words to transmit signaling information. To allow the use of widehand terminals in such situations, the initial widehand mode could he Mode 2. It has heen suggested that both hits 7 and 8 he employed here for reliable flag transmission. Corruption of the flagon bit 8 would then indicate to the terminals that their transmission channel is limited to 56 khi t/s.

Data-Speech Multiplexing
In many conference situations, i t is desirable to transmit conference-related data, such as speaker-identification information 01- document facsimile, on the established connection without interrupting the speech communication path. T h e fact that high-quality audio transmission can he maintained down to 48 kbit 's with hut minor degradation alloivs u p to 16 kbit s of data transmission. Type 2 terminals may switch from Mode 0 (64-khit s speech, no data) to Mode 1 (56-khit s speech, 8-khit! s ser\,icr channel) b y exchanging flags. It is envisaged that this service channel will provide terminal-to-termiii~il signaling and incorporate a frame alignment signal (FAS) o f 8 hits '80octt.t frame, a bit-rate~illocation signal (BAS) o f 8 bits, 80 octet frame, and u p to 6.4 khit s of data. ITnder the control of the BAS, the senice chanricl may tx expanded in increments of 8 khit s b y stealing additional bits from the speech chaiinrl. If more than 16 khit s are to be devoted to d:rta transmission, the benefits of ividehand audio heconic. marginal. Additional modes of audio transmission, e.g.. narrou.hand audio within 32 khit s (possibly C.721) 01even 16 khit) s, can be readily defined and \vould allow the capacity of the service channel to increase to 52 and 48 k hi t s , I-espect i ve 1y . Sprerh t ran sin i ssion , ;I1though s t i 1I quite intelligible, would I>r of lesser qualit\..

Communication between Narrowband and Wideband Terminals
To a1low ;I utlio conferences het w e m participants, soin e ha 1.i11g w i deha nd term i ria 1s , o t hers na rrotvba n d terminals, con\wsion of narroivlxirid signals to widehand representation and widehand signals t o narrowhand represen t;i t ion is required. Narroivha nd term i na 1s c;i n no t reproduce the high-band components o the widehand f signal; this permits their derivation from only the lowtxind component by filtering and PCM coding. However, generating a wideband signal using a QMF synthesis filter with only low-band a n d n o high-band input results in audible high-frequency distortion due to the uncanceled aliasing product. It is preferahle, thereHigher sub-band (8 kHz sampling alternate filter (high

(G.7113

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fore, to generate a pseudo-high-hand signal from the narrowband i n p u t arid use i t later to cancel the aliasing products of the loiv-hand signal. T~vo alternative procedures suggest themselt,es for the nari-owhand~ wideband conversion. T h e first and more straightforward is to upsample the uniform PCM representation t o 16 kHz, loivpass the result t o eliminate the 4-8-kHz aliasing component, arid thcn \videhand a encode the result ; i s if i t ~ v c r e normal widehand signal. A second 111-ocedure simpler and avoids the need for a is new lo\v-p;iss filter design. As shown in Fig. 8, i t generates 21 lower suhhand signal by a series of tlvo Q M F high-piss opera tioiis on the aliased narrowhand signal. I t also generates an artificial high-hand signal ljy ; series I of high- and low-pass operatiom on the same aliased signal. \l.'hen the t l v o suhtxirid cornpotieIits are p;issed through Q M F synthrsis in : ~ ~ i t l c h a ~ i d I terminal, ; I naiiowhand signal is heard Lvith no additional noise. <:on ferencc bridges combine several input signals and ma) ti-ansmit different o u t p u t signals, depending on whether the Ixirticulai port is consitlered acti1.e (currently speaking) o r silent. FOI witlehancl audio bridges, i t appeai-s preferable to c~omhiriethe l o i v - and high-hand (oniponents of the input signals f r o m the several ports separately, ;is this avoids thc accumulation o f delays due to Q M F analysis and synthesis :it conference bridges. T o ;ichic\,r hest quality when mixing narrowband and ~vidchand inputs, n;irrowlxind inputs should first be c.on\.erted to wideband form. T h e all-widehand bridge i n a y then ern ploy signa 1 coin hina t ion 1ogic ana logou s to that found in nai-roivhand bridges, hut implement i t sepal-atrlyfor the 1 0 ~ 'and high-hand signal components. -

References
[ I ] \V. R. Daumrr, P. hlermelstein, X. Maitre, and I . T o k i m r v a , "O\,rrvie\vo f the ADPCM coding algorithm," Cortf. Record Globrcom '81, Atlatita, GA, pp. 2 3 . l . l 23.1.4, 1984. [2] E. Hiirnmel, "The CCITT," ZEEE Coinmun. .&Ing., vol. 23, 110. I , pp. 8-11, 1985. [3] IEEE: (;loha1 ~ I ' e l e c o m m i ~ n i c a t i o ~ ~ s C:onfrretltc--(;lol)ecorn `86, Houston, ~1.X. Papers 17.1 thtough 17.5, 1986. [-11 T. Nishitani, I. Kui-oda,M. Satoh, T. Katoh, :iricl Y . Aoki, "A C:C:I?`T stantlard 3 2 k b s ADPCM I.SI c o c k , " ZEEE Tmn.\. , 4 ~ oust, Speech Sig-.Pro(.., \ol. ASSP-35, pp. 219-225, 1987. [ 31 R. E:. (:rochiere and J. Flan;rgan, "Curt ctit pu-spcc t i v r in digital sperch," IEI.:E C o m m i o t . itlag,, v o l . 21, no. I , pp. 32- 10, 1S83. [6] G . \Villiatiis ant1 H. Suytlcrhoud, "Suhj(~( tive pctfot tiiaii(r r\~aI~~:itioii 32 kl) s ADP(:hl iilgorithm," it] of ihc t . Corif.-C;lobecom `81?
[ 71 R. E. <:roc-liicrc, S . .A. \Vchber, ;incl J. L. Flanagan, "Digital coding 01 ~precli sub-hancls," BPI/ S y \ l . Tec I t . in J., pp. 1069- 1085, 1976. [ 8 ] N . S. J a y a n t and P. Noll. Zligitcil Codiit,q of li'rcuefoi-rrz.\. F,tiglc\vood CIiJfs, N J : Prentice-Hall, 11. 306, 1984. [9]D. Millar a n c l P. hleriiielstcin, "Prevention of prrdictoi mistrac king in ADPChl todrrs," Pro(, I E E E Z r i t . Cortf. Corn ni i t ti. -ZCC `81,A m s tcrtla in, pp. 1508- 15 12 , 1 98-1. [ I O ] D. J. Gootiinan. "Eml~cddetl DPCXI f o t - I'ariahlr Bit Kate ~1.1-ansinissioti," ZEEE Trcitt.\. Conzinuii.. \,ol. COM-28, pp. 10 10-10-l(i. 1980.

Concluding Remarks
The new widehand coding stantlard represents a n iniportant advance in two aspects: first, i t improves the quality of audio o n the telephone netlvork; second, i t provides for a n audio-associated data channel to carry conversation- or conference-related data. Its deployment is currently limited t o locations accessihle hy .i6- or 64khit, s digital loops. However, since there is much current interriational interest in digital networks such a s I S 1 I , the pen` t r:i t i on of end - t o-end dig i t ;i 1 con nec t i 1.it y )V will proha1)ly increase rapidly, and the cost of digital transmission will decrease simul tancously. T h e adoption of the new standard is timely and should not only prevent the pro1ifem t i on of incorn pat iblc cotli rig techn iq u c s , hut also help i 11 m i king cost -cf fec t i1.e premium-quali t y audio term i na 1 s ra pi dl y :i\.:i i la hle .

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