WO2001035394A2 - Integrated voice and data transmission based on bit importance ranking - Google Patents
Integrated voice and data transmission based on bit importance ranking Download PDFInfo
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- WO2001035394A2 WO2001035394A2 PCT/US2000/041953 US0041953W WO0135394A2 WO 2001035394 A2 WO2001035394 A2 WO 2001035394A2 US 0041953 W US0041953 W US 0041953W WO 0135394 A2 WO0135394 A2 WO 0135394A2
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- signal
- bits
- integrated
- bit
- voice
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- 230000005540 biological transmission Effects 0.000 title description 8
- 238000000034 method Methods 0.000 claims abstract description 23
- 238000004891 communication Methods 0.000 claims description 17
- 238000012545 processing Methods 0.000 claims description 3
- 238000001914 filtration Methods 0.000 claims 1
- 230000001172 regenerating effect Effects 0.000 abstract description 4
- 238000010586 diagram Methods 0.000 description 9
- 230000010354 integration Effects 0.000 description 5
- 238000001228 spectrum Methods 0.000 description 5
- 230000003595 spectral effect Effects 0.000 description 3
- 238000013459 approach Methods 0.000 description 2
- 230000001413 cellular effect Effects 0.000 description 2
- 238000012986 modification Methods 0.000 description 2
- 230000004048 modification Effects 0.000 description 2
- 238000000926 separation method Methods 0.000 description 2
- 230000006835 compression Effects 0.000 description 1
- 238000007906 compression Methods 0.000 description 1
- 239000002699 waste material Substances 0.000 description 1
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/80—Generation or processing of content or additional data by content creator independently of the distribution process; Content per se
- H04N21/81—Monomedia components thereof
- H04N21/8106—Monomedia components thereof involving special audio data, e.g. different tracks for different languages
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/20—Servers specifically adapted for the distribution of content, e.g. VOD servers; Operations thereof
- H04N21/23—Processing of content or additional data; Elementary server operations; Server middleware
- H04N21/235—Processing of additional data, e.g. scrambling of additional data or processing content descriptors
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
- H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
- H04N21/435—Processing of additional data, e.g. decrypting of additional data, reconstructing software from modules extracted from the transport stream
Definitions
- This invention relates to the transmission and receiving of analog and data traffic.
- the invention relates to the integration of a digitized analog signal with a separate data signal to provide an integrated signal.
- the invention relates to the integration of voice and data traffic in radio communication systems. More specifically, the invention relates to the integration of voice and data signals in cellular radio systems.
- the spectral characteristics of human speech allow its separation into three bands that have different degrees of usefulness in the reproduction of the speech signal at the receiver: DC to 300 Hz, 300-3200 Hz, and 3200- 4000 Hz.
- the first and third of these bands are relatively unimportant to the communication of speech.
- an early method used, for example, a set of filters to shoehorn a low-bit-rate' data channel into either the DC-300 Hz band or the 3200-4000 Hz band, which operated simultaneously with the transmission of speech over the 300-3200 Hz band.
- More recent proposals suitable for use in a radio communication system such as a cellular telephone system have advocated at least two general approaches: (1) collecting scraps of unused radio frequency ("RF") spectrum (rather than audio spectrum as mentioned above) in one way or another, and using these scraps to overlay a data communication capability; and (2) devoting RF spectrum in an orderly way to providing a set of channels for data communications alongside the set of channels for digitized speech.
- RF radio frequency
- a system for integrating an analog source with a separate data source to provide an integrated signal, transmitting the integrated signal, receiving the integrated signal and regenerating the analog signal and data source.
- an analog voice signal is converted to a digital signal comprising a plurality of bands. The energy of each band is approximated and compared by means of an importance gauge to a predetermined threshold level in order to identify high energy bands.
- the information in the high energy bands is compressed and integrated with a separate data signal to provide an integrated voice-data signal that is transmitted as a radio signal on a single channel.
- a receiving station essentially reverses the process to regenerate the voice signal and data signal.
- an object of this invention is to provide a system for integrating a digitized analog signal with a separate data signal to provide an integrated signal.
- Another object is to provide a system for integrating a voice signal with a data signal to provide an integrated signal.
- Another object is to provide a system for transmitting an integrated signal.
- Still another object is to provide a system for receiving an integrated signal.
- Another object is to provide a system for regenerating an analog signal and a separate data signal from an integrated signal.
- a further object is to provide a system for regenerating a voice signal and a separate data signal from an integrated signal.
- FIG. 1 illustrates a block diagram of a transmitter known in the prior art.
- FIG. 2 illustrates a block diagram of a frame known in the prior art.
- FIG. 3 illustrates a block diagram of a receiver known in the prior art.
- FIG. 4 illustrates a block diagram of a communication system including a transmitter and a receiver known in the prior art.
- FIG. 5 illustrates a block diagram of a frame known in the prior art.
- FIG. 6 illustrates a block diagram of the overall functional logic arrangement of a communication system including a transmitter and receiver according to one embodiment of the invention.
- FIG. 7 illustrates a block diagram of a transmitter according to one aspect of the invention.
- FIG. 8 illustrates a flow chart depicting the logic steps involved in integrating a voice signal with a data signal according to one aspect of the invention.
- FIG. 9 illustrates a block diagram of a receiver according to one aspect of the invention.
- FIG. 10 illustrates a flow chart depicting the logic steps in processing an incoming integrated voice-data signal according to one aspect of the invention.
- FIG. 1 1 illustrates a block diagram of a receiver according to one aspect of the invention.
- Sub-band Speech Coding of the Prior Art The purpose of source coding is to remove redundancy from a waveform that is to be digitized and communicated across a bandwidth-limited channel.
- One well established source-coding method is sub-band coding.
- the terms "speech”, “voice”, and “sound” will hereafter be regarded as equivalent terms.
- a microphone 10 picks up sound energy in the form of an analog waveform.
- the analog waveform is sampled digitally at the rate of 8,000 samples per second by an analog to digital converter 20, each sample having perhaps 16 bits of resolution.
- the resulting outpouring of samples is grouped into a sequence of binary blocks. There are 128 samples per block, 16 bits per sample, for a total of 2048 bits per block. Each block has duration of 16 milliseconds ("ms")
- the binary blocks are put through a bank of parallel bandpass filters at 30 to provide a set of band-limited signals.
- a bank of eight bandpass filters may be employed as depicted in FIG 1
- the first filter can be configured to pass 0-500 Hz
- the second filter 500-1000 Hz ...
- Quadrature mirror filters may be used to cancel sample aliases when the receiver reassembles the bands into a unified signal
- the output from bandpass filters 30 is compressed or re-coded at a lower rate by means of a shared quantizer
- the output from the quantizer 40 is modulated at 50 in order to generate an output signal 60
- FIG 2 shows how the block of
- the spectral characteristics of human speech allow its separation into a plurality of bands that have different degrees of usefulness in the reproduction of the speech signal at the receiver
- the exemplary coder of FIG I employs eight bands, which are encoded unequally by the quantizer
- the re-coding is deliberately unequal to reflect the relative importance of the various outputs from the bandpass filter bank 30
- the 512 bits might be distributed as depicted in Table 1 and FIG. 2
- band 4 is encoded with 7 bits per sample whereas the contribution of band 8 is encoded with only 2 bits per sample, see Table 1
- the unequal bit coding exploits the relative importance between the various outputs from the filter bank 30 and helps to ensure that the voice signal is encoded faithfully within the constraint of 512 bits per block
- FIG. 3 which illustrates a receiver 5' known in the prior art
- the output from transmitter 5 of FIG. 1 is received as an incoming signal 60'.
- the signal 60' is demodulated at 50' and directed to a bank of parallel bandpass; filters at 30'.
- a dequantizer 401 followed by a serializer 90 processes the output from the filter bank 30'.
- the output from the serializer 90 is directed through a digital/analog ("DIX') converter 20' in order to reproduce an analog waveform that is directed to a speaker 10'.
- DIX' digital/analog
- the receiver 5' must know a priori how the transmitter 5 apportioned bits to each band in order to ensure a faithful reproduction of the voice signal at speaker 10' Consequently inefficiencies in the transmission of the voice signal from the transmitter 5 of FIG. 1 are carried over to the receiver 5' of FIG. 3.
- FIG. 4 which depicts a communication system known in the prior art, includes a transmitter 5 and receiving apparatus 5'.
- the transmitter 5 incorporates a sub- band coder which adjusts the allocation of bits to each band as the nature of the speech signal varies with time from voiced to unvoiced, high pitch to low pitch and so on.
- the energy of each band from the filter bank 30 is approximated and communicated to a bit allocation algorithm 120.
- the bit allocation algorithm 120 allocates proportionally more bits to bands where substantial energy is detected. Since the receiver 5' of FIG. 4 can no longer rely on a priori knowledge of the transmitter, information on the allocation of bits by the bit allocation algorithm 120 is stored in a bit allocation information field 130 (see FIG. 5).
- the bit allocation field 130 is incorporated in each frame 140.
- Each frame 140 may be formatted as shown in FIG. 5.
- a complimentary bit allocation algorithm 120' (FIG. 4) reads each bit allocation field 130 in each received frame 140 in the incoming signal 60'.
- the bit allocation algorithm 12' configures an inverse dequantizer 40' in response to the contents in each bit allocation field 130 in order to correctly reconstruct the voice signal.
- the reconstructed voice signal is serialized at 90 and converted into an analog signal by the D/A converter at
- the bit allocation field 130 may contain a literal map of the bits that Mow in the frame 140.
- the bit allocation field 130 may contain a coarse version of the information upon which the transmitter 5 relied when allocating the bits to be sent, so that the receiver 5' and the transmitter 5 might have the same information and act upon it in the same way thereby eliminating the need for a literal map.
- a coarse approximation of the energy in each sub-band might be sent in the bit allocation field 130.
- the bit allocation field 130 may contain a coarse spectral profile like that described in U.S. Patent Number 4,644, 108. Coders in the prior art often waste bandwidth. The unimportant bits are often devoted to increasing the coding fidelity of the voice signal regardless of actual need.
- the unimportant bits are used primarily to code at least one separate data signal thus permitting an analog signal that has been sampled or digitized and at least one data signal to be communicated in an integrated fashion to a receiver.
- the invention is described in terms of speech or voice signals, the invention applies as well to any digitized analog signal, including but not limited to image, video, and FAX signals, as well as to digitized music including music encoded according to the NT3 standard, and also to high-bit rate signals in a native digital mode subject to compression such as 16-bit PCM signals.
- the type of source coder used in the invention is not critical.
- the type of source coder may include, but are not limited to: a sub-band coder, a vocoder, a transform coder, and a linear-predictive coder.
- An important aspect of the invention is the integration of a digitized voice signal and at least one data signal by means of importance bit ranking.
- the present invention employs bit-importance ranking to identify extra bits in the voice signal, which extra bits are used to carry a separate data signal.
- an analog voice signal is digitized to provide a voice bit stream.
- the importance of each bit in the voice bit strewn is determined against a predetermined threshold in order to identify each voice bit that is expendable verses each voice bit that is important.
- Each expendable bit is replaced with a data bit to provide an integrated voice-data bit strewn.
- the integrated voice-bit stream is modulated to provide a modulated signal.
- the modulated signal is transmitted to a receiver.
- the integrated voice-data bit stream incorporates at least one bit allocation field to enable the receiver to separate the voice signal and data signal from the integrated voice-data signal.
- FIG. 6 depicts a communication system 200, according to one embodiment of the invention, including a transmitter 210 and receiver 210 * .
- a first source 215, here an analog voice source is encoded by a source coder 220 of the present invention to provide a voice bit stream.
- the importance of each bit in the voice bit stream is analyzed by an importance gauge 230 such as a comparator.
- the terms "importance gauge” and “comparator” will hereafter be regarded as equivalent terms.
- the importance gauge 230 compares the importance of each bit against a predetermined threshold. Those bits in the voice bit stream that fall below the threshold are deemed expendable.
- the importance gauge 230 may be absolute, as just described, meaning that the importance gauge 230 judges the importance of the bits of the digitized analog signal with regard to a threshold but without regard to the nature of the data signal. More generally, the importance gauge may be relative, meaning that the importance gauge 230 compares the importance of the bits of the digitized analog signal with the importance of the data signal. For example, if a buffer holding the data signal is approaching its maximum capacity, then the data signal may be given priority over the least important bits of the digitized analog signal irrespective of a threshold. It should be understood that in this aspect of the invention the least important bits of the digitized analog signal qualify as expendable bits. In other cases, the data signal may have enhanced priority when it has been held without opportunity for transmission past an expiration time.
- the data signal may have enhanced priority relative to the bits of the digitized speech signal when the data signal has an urgent purpose such as the communication of a network management alarm or network congestion relief instruction.
- the comparator 230 configures a multiplexor 240 of the present invention, wherein the output from the source coder 220 and the signal source 225 are integrated to provide an integrated voice-data signal.
- the integration takes the form of replacing expendable bits with bits from the second signal source 225.
- the integrated voice-data signal is modulated by a modulator 250 of the present invention to provide an outgoing integrated signal 260.
- the modulated output signal 260 is received as an incoming integrated signal 2601 with respect to the receiver 210'.
- the signal 260' is demodulated at 250' and the comparator 230 determines the importance of each bit.
- the comparator 230 examines the bit-allocation information 130 (see FIG. 5) in order to determine the importance of each bit in the demodulated signal.
- the comparator 230 configures a demultiplexer 240'.
- the demultiplexer 240' separates the demodulated signal into a voice bit stream and a data bit stream.
- the voice bit stream is decoded by a source decoder 220' and forwarded to sink #1 at 215'.
- the data bit stream is directed to sink #2 at 225'.
- a speech source 215 is digitized by means of an A/D converter 218 and separated into bands by a filter bank 222.
- a bit allocation gauge 280 approximates the energy of each band.
- a shared quantizer 235 processes the bands.
- the bit allocation gauge 280 communicates with a bit allocation algorithm 290, an importance gauge 230 and a multiplexor 240.
- the bit allocation algorithm 290 configures the shared quantizer 235.
- the multiplexor 240 receives the output of the shared quantizer 235 and is configured by data received from the importance gauge 230 and the bit allocation gauge 280. When configured to do so, the multiplexor 240 integrates the output of the shared-quantizer 235 with digital data from a digital source 225.
- the output of the multiplexor 240 is modulated at 250 to provide an outgoing integrated signal 260.
- a voice signal at 215 is digitized at 218 to provide a digitized voice signal.
- the digitized voice signal is passed through a set of parallel band-pass filters
- a bit allocation gauge 280 approximates the energy of each band.
- the approximate energy of each band is forwarded to the importance gauge 230.
- the importance gauge 230 compares the energy of each band against a predetermined threshold and thereby identifies bits that are important versus bits that are expendable. Alternatively, the energy of each band is compared against a universal threshold.
- bits are allocated conventionally to the quantizer 235 according to a bit allocation algorithm at 290.
- all available bits are toward the conventional encoding of the voice signal from the analog speech source 215 (the first source).
- the multiplexor 240 is appropriately configured.
- the analog speech source 215 is sampled and digitized by the A/D converter (not shown) and separated into bands at 218.
- the energy of each band is approximated at 280.
- the energy of the lowest energy band in each block is compared to the threshold at 230. If it is found, at 270, that the lowest energy band is above the threshold then all the bits (i.e. M bits) representing the block are allocated to the analog signal according to a first bit algorithm at 292.
- a second bit algorithm at 300 allocates N bits, which would otherwise have been allocated to the low energy bands, to the data signal from the second source 225.
- the second bit algorithm then allocates M-N bits at 310 to the high-energy bands.
- the speech and data signal are encoded at 320, multiplexed and transmitted along with energy estimates or hard map at 330.
- the energy estimates for each band or hard map may be encoded in the bit allocation field 130 of each frame 140 at 330.
- FIG. 8 suggests that the allocation of N bits to the data signal and M-N bits to the voice signal occurs in series, it should be understood that the allocation of N bits to the data signal 225 and M-N bits to the voice signal 215 may alternatively be carried out in parallel.
- the N-bit-allocation for that band is forced to zero, and bits are allocated to other bands according to a second bit allocation algorithm that allocates M-N bits for encoding the speech signal from the analog source 215 and N bits towards encoding the data signal from the second source 225.
- the multiplexor 240 is configured by the comparator 230 to integrate bits from the second source 225 into the nominal voice bit stream in order to provide an integrated voice-data bit stream.
- expendable bits in the voice signal are replaced with bits from a separate data signal to provide an integrated voice-data signal.
- the integrated voice-data signal is modulated at 250 (FIG. 7) to provide an outgoing integrated signal 260.
- the importance gauge 230 might determine the importance of the bits in each band by comparing the approximate energy of each band against a predetermined threshold. For example, if the energy of band 2 were low, i.e. below the threshold, then all of the bits in band 2 would be deemed to be unimportant and band 2 would not be encoded. Thus the bits in the 512-bit block within each frame 140 that would normally be allocated to band 2 would be available to encode bits from a second signal source 225.
- the energy estimates for each band or hard map may be encoded in the bit allocation field 130 of each frame 140, see FIG. 5.
- the bit allocation field 130 of each frame 140 is read by the receiver 210' (see FIG. 9) and used to configure the de-multiplexor 240' and inverse quantizer 235'. In principle, if the bit allocation field is used to explicitly store a hard map of each 512-bit block in each frame
- the bit allocation field may be read by the bit allocation algorithm 290' and used to configure the de-multiplexor 240' and inverse quantizer 235' without requiring the functionality of a comparator 230 at the receiver 210', see FIG. 1 1.
- the transmitter 210 would have performed the logical steps to integrate the voice and data signals along with transmitting the information necessary to separate the integrated signal 260' at the receiver 210'.
- bits are freed-up.
- the freed-up bits are not allocated to improve the fidelity of the speech signal but rather, the freed-up bits are allocated by a second bit algorithm, see FIG. 8, to carry bits from a second source 225 such as a data signal.
- the comparator 230 is not restricted to evaluating the importance of bits according to the approximated energy of a band.
- the comparator 230 may determine the importance of bits according to traffic class.
- network management or control information may be judged more important than the signal provided by band 8 of a speech coder.
- network management or control information may be regarded as more important than the lowest- energy bands without regard to a threshold.
- the comparator 230 may judge the importance of bits according to tariffing considerations.
- the comparator might be configured or programmed to judge the importance of bits according to any measurable factor or absolute factor that happens to be of interest at that time.
- the receiver 210' receives an incoming integrated signal 260' which is demodulated at 250' to provide a demodulated signal.
- the bit allocation field 130 in the demodulated signal is analyzed by the comparator 230 and compared against a threshold. If the energy of the band with the lowest energy is above the threshold then the de-multiplexor 240' is appropriately configured and M bits are allocated in the conventional way to the filter bank 222'. M bits corresponding to the voice signal are processed by the inverse quantizer 235', and serialized at 238 to provide a serialized signal.
- the serialized signal is converted to an analog signal at 218'.
- the analog signal is output at speaker 215'.
- N bits corresponding to the data signal are allocated to source #2 according to a second bit algorithm (see FIG. 10).
- M-N bits in high-energy bands are allocated to the speech source.
- the demultiplexer 240' is appropriately configured to demultiplex, the bit stream. N bits are directed to sink #2 at 225', and M-N bits to the filter bank 222'. M-N bits corresponding to the voice signal are processed by the inverse quantizer 235, serialized at
- the receiver 210' essentially reverses the process of the transmitter 210 to regenerate the voice signal and data signal.
- the incoming integrated signal 260' is demodulated to. provide a demodulated signal.
- Each bit-allocation field in the demodulated signal is examined at 360 and the lowest energy is compared to the threshold at 230'. If the lowest energy band in a 512-bit block is above the threshold at 270' then all 512 bits are allocated to the speech signal according to a first bit algorithm at 292' and are processed in a conventional manner. If the lowest energy band is below the threshold then N bits are allocated at
- M-N bits are allocated to the speech source at 380.
- the de-multiplexor is configured and de-multiplexes the demodulated signal at 240'.
- N bits are routed to sink #2 at 390.
- M-N bits are decoded at 400 and the speech signal is processed in the conventional way at 410. It should be understood that expendable bits are replaced with data bits according to demand. Periods might arise when there are no data bits available to replace the expendable bits. During these periods the information stored in the bit allocation field 130 will enable the receiver 5' to allocate bits to reconstruct the analog signal according to a 1 st bit algorithm (see FIG. 10).
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Priority Applications (1)
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AU30785/01A AU3078501A (en) | 1999-11-08 | 2000-11-07 | Integrated voice and data transmission based on bit importance ranking |
Applications Claiming Priority (2)
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US43505199A | 1999-11-08 | 1999-11-08 | |
US09/435,051 | 1999-11-08 |
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WO2001035394A2 true WO2001035394A2 (en) | 2001-05-17 |
WO2001035394A3 WO2001035394A3 (en) | 2001-12-13 |
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PCT/US2000/041953 WO2001035394A2 (en) | 1999-11-08 | 2000-11-07 | Integrated voice and data transmission based on bit importance ranking |
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Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4703480A (en) * | 1983-11-18 | 1987-10-27 | British Telecommunications Plc | Digital audio transmission |
US4899384A (en) * | 1986-08-25 | 1990-02-06 | Ibm Corporation | Table controlled dynamic bit allocation in a variable rate sub-band speech coder |
-
2000
- 2000-11-07 WO PCT/US2000/041953 patent/WO2001035394A2/en active Application Filing
- 2000-11-07 AU AU30785/01A patent/AU3078501A/en not_active Abandoned
Patent Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4703480A (en) * | 1983-11-18 | 1987-10-27 | British Telecommunications Plc | Digital audio transmission |
US4899384A (en) * | 1986-08-25 | 1990-02-06 | Ibm Corporation | Table controlled dynamic bit allocation in a variable rate sub-band speech coder |
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AU3078501A (en) | 2001-06-06 |
WO2001035394A3 (en) | 2001-12-13 |
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