PATéNT DATE
`i§9?@£
`
`PATENT M
`NUMBFR ~
`
`
`NOV 1, Qt
`
`Efimfifififi
`
`APR 1 E L583
`
`6? flfififimiflié
`
`[10
`ATTORNEY'S
` Foreign priority claimed
`Ci yes
`HO
`35 USC 119 conditions mat Dyes
`
`DOCKET NO.
`
`V°"fi°'-‘V 80¢ Ackwwledaed
`examiners Initials
`
`
`:
`1
`
`,1 ADDRESS
`
`W. 20/ /W/~/'/)6 )r‘e/‘D/‘:2/It’/,»5u.'/a Kffl
`
`
`
`
`
`
`ii
`_ U.S. DEPT. OF COMM.l PATJI TM—‘PTO-436L (Rev.12-94)
`
`
`
`" I /
`,4
`._
`=
`,_
`CLAIMS“ LLOWE’ ~
`Total Claims
`*
`
`
`
`PARTS or= APPLICATION
`FILED SEPARATELY
`NOTICE or ALLowANc§»1‘IIAILED
`
`’
`
`r
`
`
`
`I
`
`5
`I
`
`.
`
`.
`
`
`
`E1-
`
`
`
`
`DRAWING av”;
`
`
`
`Amount Due
`
`
`
`1554mm"
`
`‘A
`
`same 331143293 I
`P 'mary Examiner
`PREPARED» FOR ISSUE
`
`,
`
`BATCH
`NUMBER
`
`ISSUE
`
`
`
`
`
`
`WARNING: The information disclosed herein may be restricted. Unauthorized disclosure may be prohi ~
`
`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 1113
`RPX Exhiit 113
`RPX v. DAE
`RPX V. DAE
`
`
`RPX Exhibit 1113 - Page 1
`RPX Exhibit 1113 - Page 1
`"
`
`n—A 1-u—\
`
`Pm PT0436A
`(Rev. 8/92)
`
`

`
`
`“'
`VV
`'
`Ex
`‘
`'
`“a v.
`
`t‘
`
`I
`
`t
`
`
`
`
`
`PfiTEN:'|" fiPPLIEfiT IUN
`
`\mmnmuwunxxI\uu11n\\mn\\u
`
`08540818
`
`
`
`1
`
`3
`
`V
`
`1
`
`
`Date
`’ Received
`9"
`‘
`‘ Mailed
`
`
`Date
`or
`Entered
`Counted
`
`C
`
`32.‘
`
`
`
`-
`
`’
`(FRONT)
`
`.______V
`
`‘
`A
`RPX Exhibit 1113 - Page 2
`RPX Exhibit 1113 - Page 2
`
`

`
`,:f,3,,W,,»., - ~
`
`'
`
`,
`
`,. ,,
`
`/5;
`
`PATENT APPLICATION SERIAL
`
`oqgsaosib
`
`U.s. DEPARTMEQT OF C0MEKCE:.;‘AE
`PATENT AND
`OFFICE
`'
`FEE RECORD SHEET
`
`nw
`
`PTO-1556
`<5/87)
`
`
`
`
`
`RPX Exhibit 1113 - Page 3
`RPX Exhibit 1113 - Page 3
`
`

`
`BAR CDDE LABEL
`
`|||||l|||||||||||Illilllllillillllllililll||||||||
`
`seam Numagn
`
`O8/540,818
`
`FILING DATE
`
`10/11/95
`
`PHILIPPE FERRIERE, REDMOND, WA.
`
`**coN1~INuING DATAw*vc***w**~.\»**~k**g*a3'a**
`VERIFIED
`
`**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_xed1hereto 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
`
`,
`
`i
`
`Date
`
`Certifying Officer
`
`RPX Exhibit 1113 - Page 4
`RPX Exhibit 1113 - Page 4
`
`

`
` » V-n.~-o-is
`
`
`.w
`'3‘!
`
`1
`
`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.,
`
`Large Entity Status [x]
`
`Small Entity Status []
`
`Date: OCT.
`
`ll
`
`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)
`
`E6-795356 /91 us
`
`Date:
`
`061'. H, V151?
`
`Lewis C. Lee
`
`24
`
`25
`
`La: &. HAYES, PLLC
`
`1
`
`I0lI95l332 C:lL&HlMS‘1l069US. TRJ.doc
`
`RPX Exhibit 1113 - Page 5
`RPX Exhibit 1113 - Page 5
`
`

`
`
`
`.03ME(:1‘iO8é§p%1\§') '4 ms
`
`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 1113 - Page 6
`RPX Exhibit 1113 - Page 6
`
`

`
`wow 2
`
` ‘ECHNICAL FIELD
`
`*
`
`ls» ultra
`
`2
`
`3
`
`This invention relates to audio transmission over networks, such as cable-
`
`based and wireless networks used in telephony and computing systems.
`
`5 BACKGROUND OF THE INVENTION
`
`6
`
`7
`
`3
`
`9
`
`10
`
`11
`
`Digital audio data is transmitted over networks in many different settings.
`
`Telephone systems digitize voice and transmit digital voice data over telephone
`
`lines or cellular networks. Online service providers on the Internet can download
`
`audio files to computer users via conventional telephone or cable lines. Audio
`
`files can also be exchanged over traditional data networks, such as LANs (local
`
`area networks) and WANS (wide area networks), in a manner akin to electronic
`
`12 mail.
`
`Current
`
`implementations of audio file transmission systems involve a
`
`transmission scheme in which the audio frames carrying the digital data are a fixed
`
`size. Present day modems operate at 9.6 kbps (kilobits per second), 14.4 kbps, and
`
`28.8 kbps. The audio frames from an audio file are compressed at a bit rate for
`
`transmission over these various speed communication links.
`
`To ensure that
`
`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
`
`connected at 9.6 kbps, 14.4 kbps, and 28.8 kbps. While this rate will use most of
`
`the bandwidth available at 9.6 kbps,
`
`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
`
`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
`
`13
`
`[4
`
`15
`
`16
`
`17
`
`13
`
`19
`
`20
`
`21
`
`22
`
`23
`24
`25
`
`“N
`1’
`§“\
`
`Lee & Hayes-. PLIC
`
`1
`
`MW 30832
`
`1011951119 C:lL&HlMS1l069u.r.p02.daI
`
`RPX Exhibit 1113 - Page 7
`RPX Exhibit 1113 - Page 7
`
`

`
`penalized because they are unable to retrieve audio files of a quality commensurate
`
`with the performance of their systems.
`
`It
`
`is
`
`therefore an aspect of this invention to provide an audio data
`
`transmission system which is scaleable to the communication link to use the
`
`maximum available bandwidth.
`
`In this way, a higher quality audio transmission
`
`can be provided to better performing modems.
`
`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
`
`provided by the minimal compression rate of 8000 bits/second. For larger audio
`
`files carried over limited bandwidth channels (such as low—bandwidth telephone
`
`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
`
`file. The recipient is forced to request an audio file, await the slow transmission of
`
`the whole audio file at the minimal fixed bit rate, and then play it back.
`
`Accordingly, it is another aspect of this invention to provide an optimal
`
`quality audio streaming in which the recipient can play the audio file as it is
`
`received.
`
`20
`
`21
`
`22
`
`24
`
`SUMMARY OF THE INVENTION
`
`This invention provides an audio file distribution system which permits
`
`optimal quality audio file streaming to individual customers with varying modem
`
`rates. The audio file distribution system ‘has an audio server which configures the
`
`audio files into individual audio data blocks containing a variable number of bits
`
`Lee & Hayes. PLLC
`
`2
`
`MAY! 317832
`
`I0]I95llI9 C:\L&HlMSll06%:.p02.doI
`
`RPX Exhibit 1113 - Page 8
`RPX Exhibit 1113 - Page 8
`
`

`
`of digital audio data that has been sampled at a selected input sampling rate. The
`
`number of bits of digital data and the input sampling rate are scaleable by the
`
`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
`
`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
`
`kbps, a Version of the audio data compressed at 24255 bits/s might be sent to the
`
`receiver.
`
`The audio data blocks are then transmitted at the encoded bit stream bit rate
`
`to the intended recipient’s modem. A computing unit decodes the audio data
`
`blocks to reconstruct the audio file and immediately plays the audio file as each
`
`audio data block is received. There is no restriction of waiting for the entire audio
`
`file to be downloaded before playback. As a result, a customer can request an
`
`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
`
`transmitted, making the audio file distribution process more efficient and user
`
`convenient.
`
`To determine the appropriate block size of the audio data blocks, which
`
`20
`
`22
`
`23
`
`enables scaleability to a recipient’s effective modem connection speed, the audio
`
`server and recipient computing unit are equipped with an audio coder/decoder (or
`
`“codec”).
`
`The audio codec comprises a coder to encode digital samples
`
`representative of an audio input frame into a compressed format for transmission.
`
`The coder includes multiple quantizers for encoding the digital samples into the
`
`audio data blocks of various sizes, and a quantizer selector to select the appropriate
`
`one of the quantizers.
`
`Lem & ilcygx, PLLC
`
`3
`
`MS# 30832
`
`I0lI95llI9 C.'lL&H|MSl\l169n.npl72.do(
`
`RPX Exhibit 1113 - Page 9
`RPX Exhibit 1113 - Page 9
`
`

`
`~
`
`1
`
`In the illustrated implementation, the coder is configured according to the
`
`European Group Speciale Mobile (GSM) standard. This coder has nine-quantizers.
`
`Each quantizer encodes samples representative of an audio input frame consisting
`
`of 160 input audio samples into audio data blocks of a particular size associated
`
`There are nine different block sizes, one for each
`with that quantizer.
`corresponding quantizer. The block sizes differ according to a number of audio
`
`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
`
`result,
`
`the coder can output 27 possible encoded bit stream bit rate from the
`___,,.
`
`1 available permutations of nine block sizes and three sampling rates.
`
`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
`
`appropriate combination of block size and input sampling rate from the lookup
`
`table which maximizes the bandwidth of the receiving modem. The audio server
`
`then uses the selected sampling rate to generate audio samples and chooses the
`
`appropriate quantizer to encode those samples into the appropriate block size. The
`
`resulting encoded bit stream bit rate provides optimum quality for the receiving
`
`modem.
`
`20
`
`21
`
`22
`
`24
`
`According to another aspect of this invention, a communication system
`involving multiple communication units and an interconnecting network is also
`
`adapted with an audio codec which facilitates scaleable and optimal audio quality
`
`real—time communication. An initiating communication unit supplies the‘ effective
`
`bit rate of its associated modem to a responding communication unit.
`The
`responding communication unit then determines the smallest effective bit rate
`
`between the effective bit rates for the modems of the initiating and responding
`
`Lee & Huyex, PLLC
`
`4
`
`MSI4 30832
`
`l0l195lll9 C. ll.&H\MSIlDfi9u:.p02.doI
`
`RPX Exhibit 1113 - Page 10
`RPX Exhibit 1113 - Page 10
`
`

`
`communication units, and sends the smallest effective bit rate back to the first
`
`communication unit. From that point on, the audio codecs select an appropriate
`
`quantizer which produces the audio data blocks with the quantity of bits and input
`
`sampling rate that yield an encoded bit stream bit rate of less than or equal to the
`
`smallest effective bit rate of the modems.
`
`The audio data blocks are then
`
`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
`
`Let & Hayes; PLLC
`
`5
`
`MS# 30632 Mll95[lI9 C:lL&/ilMSl\069us.plJ2.du!
`
`20
`
`22
`
`24
`
`25
`
`RPX Exhibit 1113 - Page 11
`RPX Exhibit 1113 - Page 11
`
`

`
`further includes multiple participants 28, 29, and 30 which use services provided
`
`by headend 22.
`
`Participants 28-30 are each equipped with a corresponding
`
`computing unit 32, 33, and 34 and corresponding modem 36, 37, and 38,
`
`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
`
`interconnected to the audio server 24 via a network, represented by network cloud
`
`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
`
`telephone lines or higher—bandwidth cable networks.
`
`As an example, the audio file distribution system 20 might be an online
`
`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
`
`and downloads the audio files to the requesting computing units 32-34. As another
`
`example, the audio file distribution system 20 might be implemented as part of an
`
`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
`
`modem rate. The three most conventional modem rates are 9.6 kbps (kilobits per
`
`second), 14.4 kbps, and 28.8 kbps. Despite these different modem rates, however,
`
`the audio file distribution system 20 is capable of supplying the audio files at
`
`different bit
`
`rates which are appropriate for the receiving modem.
`
`More
`
`particularly, when requesting an audio file, the requesting computing unit transmits
`
`Lee & Hayax, PLLC
`
`6
`
`Jlrfiff 30832
`
`IOII95IlI9 C:lL&HV\/IS1l069u.r.p02.Il0l
`
`20
`
`22
`
`24
`
`-K33
`
`RPX Exhibit 1113 - Page 12
`RPX Exhibit 1113 - Page 12
`
`

`
`its present modem connection speed in terms of effective bit rate, which may be
`
`equal to or less than the modem rate. Suppose, for example, that computing unit
`
`33 requests an audio file and its modem 37 is presently operating at an effective bit
`
`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
`
`effective bit rate of 13.0‘kbps is slightly less than the maximum modem rate of
`
`14.4 kbps. This is not unusual. Often times two modems will negotiate to a
`
`slightly lower bit rate, and in cases of modem sharing, modem resources might be
`
`partly consumed by other activities thereby explaining a lower effective bit rate.
`
`The computing unit 33 sends a request for an audio file and the effective bit
`
`rate of the modem 37 to the audio server 24.
`
`In return, the audio server 24
`
`supplies a compressed version of the audio file over network 40 to computing unit
`
`33 such that a bit stream bit rate of the compressed version is less than or equal to
`
`the effective bit rate of 13.0 kbps for modem 37. For instance, the audio server 24
`
`might supply the compressed version at a bit rate of 12.955 kbps, 12.1 kbps, or
`
`11.3 kbps.
`
`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,
`
`27.5 kpbs.
`
`In this case, the audio server 24 supplies a compressed version of the
`
`audio file over network 40 to computing unit 34 such that a bit stream bit rate of
`
`the compressed version is less than or equal to the effective bit rate of 27.5 kbps
`
`for modem 37 . Here, the audio server 24 might supply the compressed version at a
`
`bit rate of 25.9 kbps, 22.6 kbps, or 15.4 kbps.
`
`The audio server 24 thereby provides a compressed Version of the requested
`
`audio file that is scaled to maximize the available bandwidth of the receiving
`
`L22 & Huyzv. FLI.C
`
`7
`
`MS!‘ 30332
`
`IOH951Il9 C:lL&H\)\/ISllfI69u.v.p02,zLu
`
`10
`
`ll
`
`20
`
`22
`
`23
`
`RPX Exhibit 1113 - Page 13
`RPX Exhibit 1113 - Page 13
`
`

`
`modern.
`
`In the examples, the audio server 24 sent one compressed version of the
`
`audio file scaled to the speed of modem 37 (i.e., S 13.0 kbps) and sent a second
`
`compressed Version of the same audio file scaled to the speed of modern 38 (i.e., s
`
`27.5 kbps). The scaleability permits delivery of variable quality audio files that
`
`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
`
`a higher performing modem) and a lower quality version to computing unit 33
`
`(which has a lower performing modem).
`
`The compressed audio file supplied from the audio server 24 consists of
`
`individual audio data blocks which contain a certain number bits of digital audio
`
`data produced at a selected sample rate. The audio data blocks have variable size
`
`depending upon the number of data bits included therein. The number of digital
`
`audio data bits and the sample rate are selected to provide an encoded bit stream
`
`bit rate that is less than or equal to the effective bit rate of the receiving modem.
`
`Upon receipt of the audio data blocks representing the compressed audio
`
`file, the computing unit 33 decodes the audio data blocks and reproduces audio
`
`sound from the audio data blocks as they are received from the audio server. The
`
`computing unit does _not wait
`
`for the entire file to be downloaded before
`
`decompressing the file; but instead plays the audio sound as the blocks are being
`
`received. The participant can thus begin listening to a requested audio file
`
`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.
`1
`
`A participant can also request multiple audio files from the audio server 24.
`
`In this case, the audio server 24 supplies a compressed version of each audio file
`
`20
`
`22
`
`23
`
`24
`
`25
`
`Lee & Hayes, PLLC
`
`8
`
`MS}? 30832
`
`IOII95ll/9 C.'lL&Hl)|/ISI\Ufi9u.r.p02.du!
`
`RPX Exhibit 1113 - Page 14
`RPX Exhibit 1113 - Page 14
`
`

`
`over network 40 to the requesting computing unit. The bit stream bit rate of the
`
`compressed versions of each audio file is less than or equal to the effective bit rate
`
`of the receiving modem. Upon receipt of the compressed versions, the computing
`
`unit decompresses the audio data blocks, mixes the results, and plays the mixed
`
`version.
`
`A
`
`The audio server 24 and computing units 32-34 are all equipped with an
`
`audio coder/decoder (or “codec”). One suitable type of codec is a time—domain
`
`codec, and more particularly, an analysis-by-synthesis predictive codec. There are
`
`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
`
`Telephone and Telegraph Consultative Committee), a European GSM (Group
`
`Speciale Mobile)
`
`standard, CTIA (Cellular Telecommunications
`
`Industry
`
`Association), and two U.S.
`
`federal standards.
`
`Table 1
`
`lists various coding
`
`standards:
`
`20
`
`22
`
`23
`
`24
`
`)
`
`LEE (2 Hayex, PLLC
`
`9
`
`A/[S11 30832
`
`](Hl95Ill9 C:\L:.l’:!llIl/ISllL|69m'.[r(l2.&Il
`
`RPX Exhibit 1113 - Page 15
`RPX Exhibit 1113 - Page 15
`
`

`
` Table 1
`
`M Standardm
`
`Rate (kbps)
`
`Coder
`
`CCITT G.71l
`
`CCITT G.721
`
`64
`
`32
`
`log PCM
`
`ADPCM
`
`CCITT G.723
`
`24, 40
`
`ADPCM
`
`CCITT G.727
`
`16, 24, 32, 40 Embedded ADPCM
`
`CCITT G.728
`
`GSM
`
`CTIA
`
`Fed. Std. 1016
`
`Fed. Std. 1015
`
`16
`
`13
`
`8
`
`4.8
`
`2.4
`
`LD—CELP
`
`RPE—LTP
`
`VSELP
`
`CELP
`
`LPC
`
`The various coders listed in Table 1 are as follows: PCM (pulse code
`
`modulation), ADPCM (adaptive differential PCM), LD—CELP (low delay code-
`
`excited linear prediction), RPE—LTP (regular pulse excitation -
`
`long-term
`
`predictor), VSELP (Vector sum excited linear prediction), CELP (code—excited
`
`linear prediction), and LPC (linear predictive coding). Fig. 2 shows a block
`
`diagram of an audio codec 50 which is based in part on the European GSM
`
`standard, but modified to perform the encoding/decoding fimctions required by
`
`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
`
`PCM format) into audio data blocks of various sizes and decodes the audio data
`
`blocks to reconstruct output audio frames of 160 audio samples.
`
`In the
`
`implementation described herein, the audio data blocks have nine different sizes of
`
`Lee & Hayes‘. PLI.C
`
`10
`
`MSW 30832
`
`ILU195]]I9 C:lL&Hli\4SIllJ617u:.p02.dar
`
`I1
`
`12
`
`13
`
`20
`
`22
`
`24
`
`25
`
`RPX Exhibit 1113 - Page 16
`RPX Exhibit 1113 - Page 16
`
`

`
`1
`
`1
`
`164 bits, 176, bits, 188, bits, 200 bits, 212 bits, 224 bits, 236 bits, 248 bits, and 260
`
`2
`
`3
`
`4
`
`5
`
`6
`
`7
`
`s
`
`9
`
`10
`
`11
`
`12
`
`bits. The difference in block sizes are caused by differing numbers of encoded
`
`signal sample bits, whereby more bits results in higher quality and fewer bits
`
`results in lower quality. However, the omitted bits for the smaller block sizes are
`
`selected such that the quality loss is negligible and not too problematic to the
`
`human auditory system. The coding scheme is RPE—LTP (regular pulse excitation
`
`-lor1g—term predictor).
`
`The audio codec 50 includes an RPE-LTP coder 52 and an RPE—LTP
`
`decoder 54. The RPE-LTP coder 52 comprises a preprocessor 56, an LPC (linear
`
`predictive coding) analyzer 58, a short term analysis filter 60, a long term predictor
`
`filter 62, and an RPE coder 64. The function of all RPE—LTP coder components
`
`
`other than the RPE encoder 64 are standard, and will not be described in detail.
`
`13 Rather, a summary of the functions are provided. A more detailed presentation of
`
`14
`
`15
`
`16
`
`17
`
`18
`
`19
`
`these components is described in the ETSI-GSM Technical Specification entitled
`
`“GSM Full Rate Speech Transcoding”, GSM 06.10, Version 3.2.0, which is hereby
`
`incorporated by reference.
`
`V Preprocessor 56 receives an input audio frame consisting of 160 signal
`
`samples that are sampled at
`
`three different
`
`input sampling rates of 18000
`
`samples/second, 11025 samples/second, and 22050 samples/second. For the 8000
`
`20 Hz sampling rate, the 160 input samples represent 20 ms of audio. For the 11025
`
`21 Hz sampling rate, the 160 input samples represent 14.5 ms of audio. Finally, for
`
`22
`
`23
`
`24
`
`the 22050 Hz sampling rate, the 160 input samples represent 7.25 msof audio.
`
`The preprocessor 56 produces an offset-free signal that is then subjected to a first
`
`order pre-emphasis filter, such as a FIR (Finite Impulse Response) filter.
`
`Lee&Hayes‘.PLLC
`
`’7
`I
`( &’
`
`1 1
`
`MS#30832
`
`IfIIl95lll9C:lL&HlMSl\069u:'.p02.daI
`
`RPX Exhibit 1113 - Page 17
`RPX Exhibit 1113 - Page 17
`
`

`
`.
`
`A
`
`1
`
`2
`
`3
`
`4
`
`5
`
`6
`
`7
`
`3
`
`9
`
`10
`
`11
`
`12
`
`13
`
`14
`
`15
`
`.
`
`The LPC analyzer 58 analyzes the 160 samples to determine coefficients for
`
`use in the short term analysis filter 60. The LPC analyzer 58 performs such tasks
`
`as segmentation of the audio frame, autocorrelation, calculation of reflection
`
`coefficients using the Schur recursion algorithm, transformation of the reflection
`
`coefficients into log area ratios LARS, and quantization and coding of the LARs.
`
`The short term analysis filter 60 filters the same 160 samples to produce a short
`
`term residual signal. The short term analysis filter 60 performs such tasks as
`
`decoding the LARs from the LPC analyzer 58, interpolating the decoded LARS to
`
`avoid spurious transients which may occur if the filter coefficients are changed
`
`abruptly,
`
`transforming the LARs into reflection coefficients, and short
`
`term
`
`analysis filtering.
`
`The audio frame is divided into four sub-frames, with each sub-frame
`
`having forty samples of the short
`
`term residual signal. The sub-frames are
`
`processed blockwise by the long term predictor filter 62 and RPE encoder 64.
`
`Each sub-frame is initially passed to the long term predictor (LTP) filter 62.
`
`15 Before processing the sub-flame, LTP parameters used in the LTP filter 62 are
`
`17
`
`13
`
`19
`
`20
`
`21
`
`22
`
`23
`
`24
`
`25
`
`estimated and updated using the current sub—frame and a stored sequence of the
`
`120 previous reconstructed short term residual samples. These LTP parameters
`
`include LTP lags N and LTP gains b.
`
`A segment of forty long term residual signal samples is obtained by
`
`subtracting forty estimates of the short term residual signal from the short term
`
`residual signal itself. The resulting segment of forty long term residualsamples,
`
`designated as “e,” is fed to the RPE encoder 64 for compression. The RPE
`
`encoder 64 encodes the long term residual samples into a compressed format for
`
`___...__......
`transmission. The compressed format contains the RPE parameters which include
`
`l /7
`I K5)
`1
`l
`
`Lee&HayAs,PLLC
`
`‘
`
`ll/{Sit 30832
`
`lUIl95lIl9C.‘lL&HW1S]\069lu.p02.duI
`
`RPX Exhibit 1113 - Page 18
`RPX Exhibit 1113 - Page 18
`
`

`
`a si
`
`amples xm, a maximum of the samples xmax, and a grid position M, as
`
`will be described in more detail below.
`
`The RPE encoder 64 also produces a segment of forty samples of the
`
`quantized version of a reconstructed long term residual signal, designated as “e’,”
`
`and sends the samples back to the LTP filter 62. The forty quantized samples of
`
`the long term residual are added to the previous sub-frame of forty short term
`
`residual signal estimates to produce a reconstructed version of the current short
`
`term residual signal. This sub-frame of reconstructed short term residual signal
`
`samples is then fed back to produce a new sub—frame of forty short term residual
`
`signal estimates, thereby completing a feedback loop used in predictive coders of
`
`this type.
`
`The RPE parameters (xm, xmax, M) and LTP parameters (N, b) for all four
`
`13
`
`14
`
`sub-frames, along with the filter parameters (LARs), are configured into audio data
`
`blocks 66 of various sizes. These audio data blocks are then transmitted to the
`
`15 RPE-LTP decoder 54.
`
`16
`
`17
`
`13
`
`19
`
`20
`
`Fig. 3 shows a block diagram of the RPE encoder 64 in more detail. RPE
`
`encoder 64 has a weighting filter 70, an RPE grid selector 72, nine quantizers 740-
`
`743, nine corresponding inverse quantizers 760-763, and an RPE grid positioner
`
`78. The weighting filter 70 is a FIR filter that is applied to each sub-frame by
`
`convolving the forty long term residual samples e with an ll—tap impulse response.
`
`21 One suitable impulse response is provided in the above-referenced and
`
`22
`
`23
`
`24
`
`25
`
`incorporated ETSI—GSM Technical Specification. This filtering process yields a
`
`filtered signal X.
`
`The RPE grid selector 72 down-samples the filtered signal x by a ratio of
`
`three to yield three interleaved sequences consisting of 14, 13, and 13 samples.
`
`13
`
`[vmI3{)a'32 11,115,,”c..WW,s,.m....,,12..,,,.
`
`RPX Exhibit 1113 - Page 19
`RPX Exhibit 1113 - Page 19
`
`I K
`
`(/}' 1m1H.,1..,m
`
`

`
`The RPE grid selector then splits these sequences into four sub—sequences xm,
`
`where “m” denotes the position of a decimation grid. Each sub-sequence xm has
`
`thirteen RPE samples. The RPE grid selector 72 selects an optimum sub-sequence
`
`XM which has the maximum energy from among the four sub—sequences, where
`
`“M” denotes the optimum grid position.
`
`One of the quantizers 740-743 encodes the sub-sequence of RPE samples
`
`into a compressed format for transmission. More particularly, the selected sub-
`
`sequence xM(i) of thirteen RPE samples is quantized by one of the quantizers 740-
`
`743 using APCM (Adaptive Pulse Code Modulation).
`
`To perform the
`
`quantization, a maximum xmax of the absolute value |xM(i)| is selected for each
`
`sub-sequence of thirteen samples xM(i).
`
`The maximum xmax is quantized
`
`logarithmically and output as one of the RPE parameters in the audio data block
`
`66. The thirteen RPE samples of the selected sub-sequence xM(i) are then
`
`normalized by a decoded version x’maX of the block maximum, as follows:
`
`x’(i) = xM(i)/x’maX;i = 0,...,l2
`
`The normalized samples x’(i) are quantized uniformly with one of the nine
`
`quantizers 740-743. The appropriate quantizer is selected by the RPE grid selector
`
`72 depending upon the best available effective bit rate wMaxBitRate at which the
`
`receiving modem is presently operating. In this manner, the RPE grid selector also
`
`functions as a quantizer selector. The effective bit rate wMaxBitRate of the
`
`receiving mode is known by the RPE encoder prior to the quantization process.
`
`In
`
`the system of Fig. 1, for example, the participant computing units queried the
`
`10
`
`11
`
`12
`
`13
`
`14
`15
`
`16
`17
`
`18
`
`19
`
`20
`
`21
`
`22
`
`23
`
`24
`25
`
`,/If 1..m,,...,1m
`
`14
`
`151,113;
`
`,.,,,1,,,,1mm,..m..,_,m..1.
`
`RPX Exhibit 1113 - Page 20
`RPX Exhibit 1113 - Page 20
`
`

`
`operating system for the present‘ modem connection speed and forwarded this
`
`effective bit rate wMaxBitRate to the audio server.
`
`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
`
`different sizes of 164 bits, 176, bits, 188, bits, 200 bits, 212 bits, 224 bits, 236 bits,
`
`248 bits, and 260 bits depending upon which quantizer is selected. The various
`
`sized audio data blocks differ in the number of bits used to represent the thirteen
`
`normalized RPE samples for each of the four sub—frames. The smaller audio data
`
`blocks (i.e., 164-bit and 176-bit) contain fewer bits to represent the normalized
`
`RPE samples, whereas the larger audio data blocks (i.e., 248-bit and 260-bit)
`
`contain more bits to represent the normalized RPE samples. The fewer the bits
`
`results in a slightly lower quality signal, but not to an annoying or disruptive level.
`
`When the effective bit rate of the receiving modem is comparatively
`
`smaller, representing a lower performing modern, a quantizer that causes output of
`
`smaller sized audio data blocks is selected.
`
`The lower quality signal
`
`is
`
`commensurate with the performance of the receiving modem. Conversely, when
`
`the effective bit rate of the receiving modem is comparatively higher, representing
`
`a better performing modem, 21 quantizer that causes output of larger sized audio
`
`data blocks is selected. The higher quality signal
`
`is commensurate with the
`
`performance of the better performing receiving modem.
`
`In this manner, the
`
`multiple quantizers enable the RPE-LTP coder 52 to be scaleable according to the
`
`awaiting modem capabilities.
`
`The following discussion provides a specific example implementation of
`
`the nine quantizers used in an audio RPE-LTP coder that is implemented according
`
`to the European GS