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`DOCKET NO.
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`WARNING: The information disclosed herein may be restricted. Unauthorized disclosure may be prohi ~
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`by the United States Code Title 35, Sections 122, 181 and 368. Possession outside the U.
`Patent & Trademark Office is restricted to authorized employees and contractors only.
`
`
`
`RPX Exhibit 1013
`RPX Exhiit 103
`RPX v. DAE
`RPX V. DAE
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`RPX Exhibit 1013 - Page 1
`RPX Exhibit 1013 - Page 1
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`RPX Exhibit 1013 - Page 3
`RPX Exhibit 1013 - Page 3
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`BAR CDDE LABEL
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`|||||l|||||||||||Illilllllillillllllililll||||||||
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`seam Numagn
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`O8/540,818
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`FILING DATE
`
`10/11/95
`
`PHILIPPE FERRIERE, REDMOND, WA.
`
`**coN1~INuING DATAw*vc***w**~.\»**~k**g*a3'a**
`VERIFIED
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`**FOREIGN/PCT APPLICATIONS** ** * * "_* * * * *
`VERIFIED
`’
`
`GROUP AHT UNIT
`
`2603
`
`FOREIGN FILING LICENSE GRANTED 03/28/96
`
`FILJNG FEE
`RECEIVED
`
`ATTORNEY DOCKET NO.
`
`$1,778.00
`
`MS1069US
`
`LEE & HAYES
`w 1818 FRANCIS #160
`spoxamz WA 99205
`
`SYSTEM AND METHOD FOR SCALEABLE STREBMED AUDIO TRANSMISSION OVER A
`NETWORK
`
`that anne_xedihereto is a true copy frqm_ the _rgcords of the United States
`This is to certif
`Patent and Tra emark Office of the application which IS Identified above.
`By authority of the
`I
`CUMMISSWNER OF PATENTS AND TRADEMARKS
`
`,
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`i
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`Date
`
`Certifying Officer
`
`RPX Exhibit 1013 - Page 4
`RPX Exhibit 1013 - Page 4
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`IN THE UNITED STATES PATENT AND TRADEMARK OFF§8E
`Inventor...................................................................'...........
`....................
`.. Phkippe Ferriere
`
`Microsofi Corporation
`Applicant........... .L........ ..
`
`...................... ..MSl-069US
`............
`Attomey‘s Docket No.
`.
`System and Method for Scaleable Streamed Audio Transmission Over a Network
`Title:
`
`TRANSMITTAL LETTER AND CERTIFICATE OF MAILING
`
`To:
`
`From:
`
`Commissioner of Patents and Trademarks,
`Washington, D.C. 20231
`
`Lewis C. Lee (Tel. 509-324-9256; Fax 509-928-2642)
`Lee & Hayes, PLLC
` 9
`Spokane, WA 99205
`
`The following enumerated items accompany this transmittal letter and are being submitted for the
`matter identified in the above caption.
`., ye
`.,
`
`Specification—title page (with listed inventor) -plus 51 pages, including 36 claims
`1.
`6 sheets of formal drawings (Figs. 1-6)
`2
`3. Transmittal letter including Certificate of Express Mailing
`4
`Return Post Card
`
`i.,
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`Large Entity Status [x]
`
`Small Entity Status []
`
`Date: OCT.
`
`ll
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`iqcl-5‘
`
`
`
`By:
`
`'1; 4
`
`Lewis C. Lee
`Reg. No. 34,656
`
`CERTIFICATE OF MAILING
`
`I hereby certify that the items listed above as enclosed are being deposited with the US. Postal
`Service as either first class mail, or Express Mail if the blank for Express Mail No. is completed below, in
`an envelope addressed to The Commissioner of Patents and Trademarks, Washington, D.C. 20231, on the
`below-indicated date. Any Express Mail No. has also been marked on the listed items.
`
`Express Mail No. (if applicable)
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`E6-795356 /91 us
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`Date:
`
`061'. H,
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`Lewis C. Lee
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`RPX Exhibit 1013 - Page 5
`RPX Exhibit 1013 - Page 5
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`IN THE UNITED STATES PATENT AND TRADEMARK OPFICE
`
`APPLICATION FOR LETTERS PATENT
`
` o
`Transmission Over a Network
`
`Inventor(s):
`
`Philippe Ferriere
`
`ATTORNEYS DOCKET NO. MS1-O69US
`
`RPX Exhibit 1013 - Page 6
`RPX Exhibit 1013 - Page 6
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`wow 2
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` ‘ECHNICAL FIELD
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`This invention relates to audio transmission over networks, such as cable-
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`based and wireless networks used in telephony and computing systems.
`
`5 BACKGROUND OF THE INVENTION
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`6
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`3
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`Digital audio data is transmitted over networks in many different settings.
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`Telephone systems digitize voice and transmit digital voice data over telephone
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`lines or cellular networks. Online service providers on the Internet can download
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`audio files to computer users via conventional telephone or cable lines. Audio
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`files can also be exchanged over traditional data networks, such as LANs (local
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`area networks) and WANS (wide area networks), in a manner akin to electronic
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`12 mail.
`
`Current
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`implementations of audio file transmission systems involve a
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`transmission scheme in which the audio frames carrying the digital data are a fixed
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`size. Present day modems operate at 9.6 kbps (kilobits per second), 14.4 kbps, and
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`28.8 kbps. The audio frames from an audio file are compressed at a bit rate for
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`transmission over these various speed communication links.
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`To ensure that
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`transmission is possible over all three conventional speeds, the audio files are
`
`typically compressed at a bit rate of 8000 bits/second which can be sent to modems
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`connected at 9.6 kbps, 14.4 kbps, and 28.8 kbps. While this rate will use most of
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`the bandwidth available at 9.6 kbps,
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`it uses only a fraction of the available
`
`bandwidth at 14.4 kbps and 28.8 kbps. Since the file is compressed at a lower
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`quality rate of 8000 bits/second, the eventual reconstructed file has an equally low
`and fixed quality.
`The customers who use higher performing modems are
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`RPX Exhibit 1013 - Page 7
`RPX Exhibit 1013 - Page 7
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`penalized because they are unable to retrieve audio files of a quality commensurate
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`with the performance of their systems.
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`It
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`is
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`therefore an aspect of this invention to provide an audio data
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`transmission system which is scaleable to the communication link to use the
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`maximum available bandwidth.
`
`In this way, a higher quality audio transmission
`
`can be provided to better performing modems.
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`In the online services setting, conventional systems require transmission of
`
`the entire audio file (whether compressed or uncompressed) before the recipient is
`
`able to play back the audio file. The audio file is at one fixed quality, such as that
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`provided by the minimal compression rate of 8000 bits/second. For larger audio
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`files carried over limited bandwidth channels (such as low—bandwidth telephone
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`lines), the time required to download the whole audio file can take several minutes.
`
`This transmission delay is inconvenient to the recipient, particularly if the recipient
`
`is only browsing various audio files with little intent of listening to the entire audio
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`file. The recipient is forced to request an audio file, await the slow transmission of
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`the whole audio file at the minimal fixed bit rate, and then play it back.
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`Accordingly, it is another aspect of this invention to provide an optimal
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`quality audio streaming in which the recipient can play the audio file as it is
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`received.
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`SUMMARY OF THE INVENTION
`
`This invention provides an audio file distribution system which permits
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`optimal quality audio file streaming to individual customers with varying modem
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`rates. The audio file distribution system ‘has an audio server which configures the
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`audio files into individual audio data blocks containing a variable number of bits
`
`Lee & Hayes. PLLC
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`2
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`MAY! 317832
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`RPX Exhibit 1013 - Page 8
`RPX Exhibit 1013 - Page 8
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`of digital audio data that has been sampled at a selected input sampling rate. The
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`number of bits of digital data and the input sampling rate are scaleable by the
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`audio server to produce an encoded bit stream bit rate that is less than or equal to
`
`an effective operational bit rate of a recipient’s modem. For example, if the
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`modem connection speed is 14.4 kbps, a version of the audio data compressed at
`
`13000 bits/s might be sent to the recipient; if the modem connection speed is 28.8
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`kbps, a Version of the audio data compressed at 24255 bits/s might be sent to the
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`receiver.
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`The audio data blocks are then transmitted at the encoded bit stream bit rate
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`to the intended recipient’s modem. A computing unit decodes the audio data
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`blocks to reconstruct the audio file and immediately plays the audio file as each
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`audio data block is received. There is no restriction of waiting for the entire audio
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`file to be downloaded before playback. As a result, a customer can request an
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`audio file from the audio server and begin listening immediately. If the customer
`is just browsing, he/she is free to cancel the audio file before the entire file is
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`transmitted, making the audio file distribution process more efficient and user
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`convenient.
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`To determine the appropriate block size of the audio data blocks, which
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`20
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`enables scaleability to a recipient’s effective modem connection speed, the audio
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`server and recipient computing unit are equipped with an audio coder/decoder (or
`
`“codec”).
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`The audio codec comprises a coder to encode digital samples
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`representative of an audio input frame into a compressed format for transmission.
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`The coder includes multiple quantizers for encoding the digital samples into the
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`audio data blocks of various sizes, and a quantizer selector to select the appropriate
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`one of the quantizers.
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`Lem & ilcygx, PLLC
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`MS# 30832
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`RPX Exhibit 1013 - Page 9
`RPX Exhibit 1013 - Page 9
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`~
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`In the illustrated implementation, the coder is configured according to the
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`European Group Speciale Mobile (GSM) standard. This coder has nine-quantizers.
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`Each quantizer encodes samples representative of an audio input frame consisting
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`of 160 input audio samples into audio data blocks of a particular size associated
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`There are nine different block sizes, one for each
`with that quantizer.
`corresponding quantizer. The block sizes differ according to a number of audio
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`data bits contained in each audio data block. Moreover, each quantizer can be
`
`"operated to encode the samples for three different input sampling rates. As a
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`result,
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`the coder can output 27 possible encoded bit stream bit rate from the
`___,,.
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`V available permutations of nine block sizes and three sampling rates.
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`The 27 possible encoded bit stream bit rates can be stored in lookup tables
`
`at the audio server and recipient computing units. The audio server selects the
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`appropriate combination of block size and input sampling rate from the lookup
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`table which maximizes the bandwidth of the receiving modem. The audio server
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`then uses the selected sampling rate to generate audio samples and chooses the
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`appropriate quantizer to encode those samples into the appropriate block size. The
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`resulting encoded bit stream bit rate provides optimum quality for the receiving
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`modem.
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`20
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`24
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`According to another aspect of this invention, a communication system
`involving multiple communication units and an interconnecting network is also
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`adapted with an audio codec which facilitates scaleable and optimal audio quality
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`real—time communication. An initiating communication unit supplies the‘ effective
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`bit rate of its associated modem to a responding communication unit.
`The
`responding communication unit then determines the smallest effective bit rate
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`between the effective bit rates for the modems of the initiating and responding
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`Lee & Huyex, PLLC
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`RPX Exhibit 1013 - Page 10
`RPX Exhibit 1013 - Page 10
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`communication units, and sends the smallest effective bit rate back to the first
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`communication unit. From that point on, the audio codecs select an appropriate
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`quantizer which produces the audio data blocks with the quantity of bits and input
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`sampling rate that yield an encoded bit stream bit rate of less than or equal to the
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`smallest effective bit rate of the modems.
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`The audio data blocks are then
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`exchanged over the network at the encoded bit stream rate.
`
`BRIEF DESCRIPTION OF THE DRAWINGS
`
` diagammatic illustration of an audio file distribution system
`according to one aspect of this invention.
` block diagram of an audio coder/decoder (codec) according to
`
`another aspect of this invention. The audio codec is illustrated in an example
`
`implementation as an RPE—LTP codec according to the European GSM standard.
`‘ lock diagram of an RPE encoder employed in the Fig. 2 codec.
`yrgfii block diagram ofan RPE decoder employed in the Fig. 2 codec.
` flow diagram of a method for supplying audio files according to
`another aspect of this invention.
`
`Fig.
`
`' a diagrammatic illustration of a communication system according
`
`to anot er aspect of this invention.
`
`DETAILED DESCRIPTION OF THE PREFERRED EMBODHVIENT
`
`Fig. 1 shows an audio file distribution system 20 for supplying digital audio
`
`files to multiple different participants. Audio file distribution system 20 has a
`
`headend 22 with an audio server 24 and an audio file storage 26. System 20
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`Let & Hayes; PLLC
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`RPX Exhibit 1013 - Page 11
`RPX Exhibit 1013 - Page 11
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`further includes multiple participants 28, 29, and 30 which use services provided
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`by headend 22.
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`Participants 28-30 are each equipped with a corresponding
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`computing unit 32, 33, and 34 and corresponding modem 36, 37, and 38,
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`respectively. The participant computing units are illustrated as desktop computers,
`
`but can alternatively be in implemented as other types of personal computers,
`
`telephone units, set-top boxes, or other digital processing mechanisms that are
`
`capable of handling digital audio data. The participant computing units 32-34 are
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`interconnected to the audio server 24 via a network, represented by network cloud
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`40. The network 40 might be in the form of a wireless network, such as satellite
`
`and cellular phone networks, or a wire-based network, such as low-bandwidth
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`telephone lines or higher—bandwidth cable networks.
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`As an example, the audio file distribution system 20 might be an online
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`network system in which participants 28-30 dial up and request audio files from
`
`the headend 22. The audio server 24 retrieves the audio files from the storage 26
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`and downloads the audio files to the requesting computing units 32-34. As another
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`example, the audio file distribution system 20 might be implemented as part of an
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`interactive television (ITV) system in which subscribers 28-30 send requests over
`
`the TV cable to a cable headend for certain audio files for use in conjunction with,
`
`or separate fiom, video programs.
`
`For discussion purposes, the modems 36-38 each operate at a different
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`modem rate. The three most conventional modem rates are 9.6 kbps (kilobits per
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`second), 14.4 kbps, and 28.8 kbps. Despite these different modem rates, however,
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`the audio file distribution system 20 is capable of supplying the audio files at
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`different bit
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`rates which are appropriate for the receiving modem.
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`More
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`particularly, when requesting an audio file, the requesting computing unit transmits
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`RPX Exhibit 1013 - Page 12
`RPX Exhibit 1013 - Page 12
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`its present modem connection speed in terms of effective bit rate, which may be
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`equal to or less than the modem rate. Suppose, for example, that computing unit
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`33 requests an audio file and its modem 37 is presently operating at an effective bit
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`rate of 13.0 kbps. The computing unit 33 determines this effective bit rate by
`
`querying its operating system for the current connection speed of modem 37 . The
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`effective bit rate of 13.0‘kbps is slightly less than the maximum modem rate of
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`14.4 kbps. This is not unusual. Often times two modems will negotiate to a
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`slightly lower bit rate, and in cases of modem sharing, modem resources might be
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`partly consumed by other activities thereby explaining a lower effective bit rate.
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`The computing unit 33 sends a request for an audio file and the effective bit
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`rate of the modem 37 to the audio server 24.
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`In return, the audio server 24
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`supplies a compressed version of the audio file over network 40 to computing unit
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`33 such that a bit stream bit rate of the compressed version is less than or equal to
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`the effective bit rate of 13.0 kbps for modem 37. For instance, the audio server 24
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`might supply the compressed version at a bit rate of 12.955 kbps, 12.1 kbps, or
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`11.3 kbps.
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`Now suppose that computing unit 34 sends a request for the same audio file
`
`along with an effective bit rate of corresponding modem 38 which is, for example,
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`27.5 kpbs.
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`In this case, the audio server 24 supplies a compressed version of the
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`audio file over network 40 to computing unit 34 such that a bit stream bit rate of
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`the compressed version is less than or equal to the effective bit rate of 27.5 kbps
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`for modem 37 . Here, the audio server 24 might supply the compressed version at a
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`bit rate of 25.9 kbps, 22.6 kbps, or 15.4 kbps.
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`The audio server 24 thereby provides a compressed Version of the requested
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`audio file that is scaled to maximize the available bandwidth of the receiving
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`L22 & Huyzv. FLI.C
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`RPX Exhibit 1013 - Page 13
`RPX Exhibit 1013 - Page 13
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`modern.
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`In the examples, the audio server 24 sent one compressed version of the
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`audio file scaled to the speed of modem 37 (i.e., S 13.0 kbps) and sent a second
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`compressed Version of the same audio file scaled to the speed of modern 38 (i.e., s
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`27.5 kbps). The scaleability permits delivery of variable quality audio files that
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`are commensurate with the communication bandwidth. The audio server 24
`
`provides a higher quality version of the audio file to computing unit 34 (which has
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`a higher performing modem) and a lower quality version to computing unit 33
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`(which has a lower performing modem).
`
`The compressed audio file supplied from the audio server 24 consists of
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`individual audio data blocks which contain a certain number bits of digital audio
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`data produced at a selected sample rate. The audio data blocks have variable size
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`depending upon the number of data bits included therein. The number of digital
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`audio data bits and the sample rate are selected to provide an encoded bit stream
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`bit rate that is less than or equal to the effective bit rate of the receiving modem.
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`Upon receipt of the audio data blocks representing the compressed audio
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`file, the computing unit 33 decodes the audio data blocks and reproduces audio
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`sound from the audio data blocks as they are received from the audio server. The
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`computing unit does _not wait
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`for the entire file to be downloaded before
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`decompressing the file; but instead plays the audio sound as the blocks are being
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`received. The participant can thus begin listening to a requested audio file
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`immediately, and cancel the file if he/she desires to quit listening to that file and
`move onto another file. Additionally, by scaleably encoding individual blocks, the
`receiving computing unit is ensured of optimal quality audio data.
`-
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`A participant can also request multiple audio files from the audio server 24.
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`In this case, the audio server 24 supplies a compressed version of each audio file
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`RPX Exhibit 1013 - Page 14
`RPX Exhibit 1013 - Page 14
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`over network 40 to the requesting computing unit. The bit stream bit rate of the
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`compressed versions of each audio file is less than or equal to the effective bit rate
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`of the receiving modem. Upon receipt of the compressed versions, the computing
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`unit decompresses the audio data blocks, mixes the results, and plays the mixed
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`version.
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`A
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`The audio server 24 and computing units 32-34 are all equipped with an
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`audio coder/decoder (or “codec”). One suitable type of codec is a time—domain
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`codec, and more particularly, an analysis-by-synthesis predictive codec. There are
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`a variety of speech and other audio coding standards for different applications,
`
`both nationally and internationally. The standards are based upon different coding
`
`rates and employ different types of coders. The audio codecs are configured using
`
`one of the common standards, which includes versions of CCITT (International
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`Telephone and Telegraph Consultative Committee), a European GSM (Group
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`Speciale Mobile)
`
`standard, CTIA (Cellular Telecommunications
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`Industry
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`Association), and two U.S.
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`federal standards.
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`Table 1
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`lists various coding
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`standards:
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`RPX Exhibit 1013 - Page 15
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` Table 1
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`M Standardm
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`Rate (kbps)
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`Coder
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`CCITT G.71l
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`CCITT G.721
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`64
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`32
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`log PCM
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`ADPCM
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`CCITT G.723
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`24, 40
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`ADPCM
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`CCITT G.727
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`16, 24, 32, 40 Embedded ADPCM
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`CCITT G.728
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`GSM
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`CTIA
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`Fed. Std. 1016
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`Fed. Std. 1015
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`16
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`8
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`4.8
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`2.4
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`LD—CELP
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`RPE—LTP
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`VSELP
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`CELP
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`LPC
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`The various coders listed in Table 1 are as follows: PCM (pulse code
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`modulation), ADPCM (adaptive differential PCM), LD—CELP (low delay code-
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`excited linear prediction), RPE—LTP (regular pulse excitation -
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`long-term
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`predictor), VSELP (Vector sum excited linear prediction), CELP (code—excited
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`linear prediction), and LPC (linear predictive coding). Fig. 2 shows a block
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`diagram of an audio codec 50 which is based in part on the European GSM
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`standard, but modified to perform the encoding/decoding fimctions required by
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`aspects of this invention. The audio codec 50 is preferably implemented in
`software which executes on the audio server and recipient computing units. The
`audio codec 50 encodes input audio frames of 160 audio samples (8-bitor 16-bit
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`PCM format) into audio data blocks of various sizes and decodes the audio data
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`blocks to reconstruct output audio frames of 160 audio samples.
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`In the
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`implementation described herein, the audio data blocks have nine different sizes of
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`RPX Exhibit 1013 - Page 16
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`164 bits, 176, bits, 188, bits, 200 bits, 212 bits, 224 bits, 236 bits, 248 bits, and 260
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`bits. The difference in block sizes are caused by differing numbers of encoded
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`signal sample bits, whereby more bits results in higher quality and fewer bits
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`results in lower quality. However, the omitted bits for the smaller block sizes are
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`selected such that the quality loss is negligible and not too problematic to the
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`human auditory system. The coding scheme is RPE—LTP (regular pulse excitation
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`-lor1g—term predictor).
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`The audio codec 50 includes an RPE-LTP coder 52 and an RPE—LTP
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`decoder 54. The RPE-LTP coder 52 comprises a preprocessor 56, an LPC (linear
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`predictive coding) analyzer 58, a short term analysis filter 60, a long term predictor
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`filter 62, and an RPE coder 64. The function of all RPE—LTP coder components
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`other than the RPE encoder 64 are standard, and will not be described in detail.
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`13 Rather, a summary of the functions are provided. A more detailed presentation of
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`these components is described in the ETSI-GSM Technical Specification entitled
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`“GSM Full Rate Speech Transcoding”, GSM 06.10, Version 3.2.0, which is hereby
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`incorporated by reference.
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`V Preprocessor 56 receives an input audio frame consisting of 160 signal
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`samples that are sampled at
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`three different
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`input sampling rates of 18000
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`samples/second, 11025 samples/second, and 22050 samples/second. For the 8000
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`20 Hz sampling rate, the 160 input samples represent 20 ms of audio. For the 11025
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`21 Hz sampling rate, the 160 input samples represent 14.5 ms of audio. Finally, for
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`the 22050 Hz sampling rate, the 160 input samples represent 7.25 msof audio.
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`The preprocessor 56 produces an offset-free signal that is then subjected to a first
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`order pre-emphasis filter, such as a FIR (Finite Impulse Response) filter.
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`RPX Exhibit 1013 - Page 17
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`The LPC analyzer 58 analyzes the 160 samples to determine coefficients for
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`use in the short term analysis filter 60. The LPC analyzer 58 performs such tasks
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`as segmentation of the audio frame, autocorrelation, calculation of reflection
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`coefficients using the Schur recursion algorithm, transformation of the reflection
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`coefficients into log area ratios LARS, and quantization and coding of the LARs.
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`The short term analysis filter 60 filters the same 160 samples to produce a short
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`term residual signal. The short term analysis filter 60 performs such tasks as
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`decoding the LARs from the LPC analyzer 58, interpolating the decoded LARS to
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`avoid spurious transients which may occur if the filter coefficients are changed
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`abruptly,
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`transforming the LARs into reflection coefficients, and short
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`term
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`analysis filtering.
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`The audio frame is divided into four sub-frames, with each sub-frame
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`having forty samples of the short
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`term residual signal. The sub-frames are
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`processed blockwise by the long term predictor filter 62 and RPE encoder 64.
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`Each sub-frame is initially passed to the long term predictor (LTP) filter 62.
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`15 Before processing the sub-flame, LTP parameters used in the LTP filter 62 are
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`estimated and updated using the current sub—frame and a stored sequence of the
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`120 previous reconstructed short term residual samples. These LTP parameters
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`include LTP lags N and LTP gains b.
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`A segment of forty long term residual signal samples is obtained by
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`subtracting forty estimates of the short term residual signal from the short term
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`residual signal itself. The resulting segment of forty long term residualsamples,
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`designated as “e,” is fed to the RPE encoder 64 for compression. The RPE
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`encoder 64 encodes the long term residual samples into a compressed format for
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`___...__......
`transmission. The compressed format contains the RPE parameters which include
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`RPX Exhibit 1013 - Page 18
`RPX Exhibit 1013 - Page 18
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`a si
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`amples xm, a maximum of the samples xmax, and a grid position M, as
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`will be described in more detail below.
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`The RPE encoder 64 also produces a segment of forty samples of the
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`quantized version of a reconstructed long term residual signal, designated as “e’,”
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`and sends the samples back to the LTP filter 62. The forty quantized samples of
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`the long term residual are added to the previous sub-frame of forty short term
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`residual signal estimates to produce a reconstructed version of the current short
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`term residual signal. This sub-frame of reconstructed short term residual signal
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`samples is then fed back to produce a new sub—frame of forty short term residual
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`signal estimates, thereby completing a feedback loop used in predictive coders of
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`this type.
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`The RPE parameters (xm, xmax, M) and LTP parameters (N, b) for all four
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`sub-frames, along with the filter parameters (LARs), are configured into audio data
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`blocks 66 of various sizes. These audio data blocks are then transmitted to the
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`15 RPE-LTP decoder 54.
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`Fig. 3 shows a block diagram of the RPE encoder 64 in more detail. RPE
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`encoder 64 has a weighting filter 70, an RPE grid selector 72, nine quantizers 740-
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`743, nine corresponding inverse quantizers 760-763, and an RPE grid positioner
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`78. The weighting filter 70 is a FIR filter that is applied to each sub-frame by
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`convolving the forty long term residual samples e with an ll—tap impulse response.
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`21 One suitable impulse response is provided in the above-referenced and
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`incorporated ETSI—GSM Technical Specification. This filtering process yields a
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`filtered signal X.
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`The RPE grid selector 72 down-samples the filtered signal x by a ratio of
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`three to yield three interleaved sequences consisting of 14, 13, and 13 samples.
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`RPX Exhibit 1013 - Page 19
`RPX Exhibit 1013 - Page 19
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`The RPE grid selector then splits these sequences into four sub—sequences xm,
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`where “m” denotes the position of a decimation grid. Each sub-sequence xm has
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`thirteen RPE samples. The RPE grid selector 72 selects an optimum sub-sequence
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`XM which has the maximum energy from among the four sub—sequences, where
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`“M” denotes the optimum grid position.
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`One of the quantizers 740-743 encodes the sub-sequence of RPE samples
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`into a compressed format for transmission. More particularly, the selected sub-
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`sequence xM(i) of thirteen RPE samples is quantized by one of the quantizers 740-
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`743 using APCM (Adaptive Pulse Code Modulation).
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`To perform the
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`quantization, a maximum xmax of the absolute value |xM(i)| is selected for each
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`sub-sequence of thirteen samples xM(i).
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`The maximum xmax is quantized
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`logarithmically and output as one of the RPE parameters in the audio data block
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`66. The thirteen RPE samples of the selected sub-sequence xM(i) are then
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`normalized by a decoded version x’maX of the block maximum, as follows:
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`x’(i) = xM(i)/x’maX;i = 0,...,l2
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`The normalized samples x’(i) are quantized uniformly with one of the nine
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`quantizers 740-743. The appropriate quantizer is selected by the RPE grid selector
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`72 depending upon the best available effective bit rate wMaxBitRate at which the
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`receiving modem is presently operating. In this manner, the RPE grid selector also
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`functions as a quantizer selector. The effective bit rate wMaxBitRate of the
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`receiving mode is known by the RPE encoder prior to the quantization process.
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`In
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`the system of Fig. 1, for example, the participant computing units queried the
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`RPX Exhibit 1013 - Page 20
`RPX Exhibit 1013 - Page 20
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`operating system for the present‘ modem connection speed and forwarded this
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`effective bit rate wMaxBitRate to the audio server.
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`Depending upon which quantizer is selected, the audio data blocks output
`by the RPE-LTP coder 52 are different in size. The audio data blocks have nine
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`different sizes of 164 bits, 176, bits, 188, bits, 200 bits, 212 bits, 224 bits, 236 bits,
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`248 bits, and 260 bits depending upon which quantizer is selected. The various
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`sized audio data blocks differ in the number of bits used to represent the thirteen
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`normalized RPE samples for each of the four sub—frames. The smaller audio data
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`blocks (i.e., 164-bit and 176-bit) contain fewer bits to represent the normalized
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`RPE samples, whereas the larger audio data blocks (i.e., 248-bit and 260-bit)
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`contain more bits to represent the normalized RPE samples. The fewer the bits
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`results in a slightly lower quality signal, but not to an annoying or disruptive level.
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`When the effective bit rate of the receiving modem is comparatively
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`smaller, representing a lower performing modern, a quantizer that causes output of
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`smaller sized audio data blocks is selected.
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`The lower quality signal
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`is
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`commensurate with the performance of the receiving modem. Conversely, when
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`the effective bit rate of the receiving modem is comparatively higher, representing
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`a better performing modem, 21 quantizer that causes output of larger sized audio
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`data blocks is selected. The higher quality signal
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`is commensurate with the
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`performance of the better performing receiving modem.
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`In this manner, the
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`multiple quantizers enable the RPE-LTP coder 52 to be scaleable according to the
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`awaiting modem capabilities.
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`The following discussion provides a specific example implementation of
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`the nine quantizers used in an audio RPE-LTP coder that is implemented according
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`to the Europea