摘要:
Digital circuit multiplication equipment (DCME) refers to coding bit rate information included in a pseudo-speech signal. In accordance with the coding bit rate information, the DCME selects either a coded speech signal extracted by a transmission bit rate restorer or a coded speech signal including bit rate identification information added by a coding bit rate information adding section, and supplies the selected coded speech signal to a bearer line. The DCME can solve a problem of a conventional DCME in that a mismatch can take place between the actual transmission bit rate of the coded speech signal and the transmission bit rate assigned to the bearer line when providing the variable bit rate DCME with a tandem passthrough function, and therefore the correct coding bit rate information cannot be transferred to a speech decoder, bringing about serious degradation in the speech quality.
摘要:
The invention encompasses a method of preparing data for transmission, which method comprises: multiplexing data from a plurality of sources (S1, . . . Si, . . . Sn), the multiplexing comprising, for at least one source (Si), classifying the data from the source (Si) into two or more classes (C1,f.Cj, . . . Cm) according to the data's priority, and mapping data from the sources into positions in a data structure (D) according both to the class (Cj) of the data and to a further priority assigned to the source (Si) from which the data originated, the division of data into classes (C1, . . . Cj, . . . Cm) and/or the prioritization of the sources (S1, . . . Si, . . . Sn) being done according to the potential impact of transmission errors on the data; and sub-dividing the data in the data structure (D) into frames while preserving the relative prioritization of the data. The sources may be various multi-media sources. Also disclosed are a method of forward error correction (FEC) encoding the multiplexed data for transmission while maintaining the data's relative prioritization, and a method for decoding and de-multiplexing data. Apparatus for implementing these methods is also disclosed.
摘要:
A data multiplexing network is described which multiplexes a plurality of asynchronous data channels with an asynchronous data stream representing compressed voice signals and/or facsimile signals onto a single synchronous data packet stream. The single synchronous data packet stream is then transmitted by a high speed statistical multiplexer over a composite link to a second site using a modified high-level synchronous data link control protocol with an overlay of an advanced priority statistical multiplexing algorithm. The asynchronous data channels and the compressed voice channel and/or facsimile signals are demultiplexed and reconstructed for sending to other asynchronous computer terminals and to a standard telephone or facsimile analog port at the second site, respectively. PBX trunk interfaces are also provided to allow PBX's to share the composite link between sites. Communication between the first site by voice or facsimile and the second site is transparent to the users. The multiplexer efficiently allocated the bandwidth of the composite link by detecting silence periods in the voice signals and suppressing the sending of the voice information to preserve bandwidth. An advanced priority statistical multiplexer is also described which dynamically allocates composite link bandwidth to both time-sensitive and non-time-sensitive data to maximize data throughout efficiency and quality while simultaneously reducing multiplexer processing overhead
摘要:
A system and method for communicating information signals using spread spectrum communication techniques. PN sequences are constructed that provide orthogonality between the users so that mutual interference will be reduced, allowing higher capacity and better link performance. With orthogonal PN codes, the cross-correlation is zero over a predetermined time interval, resulting in no interference between the orthogonal codes, provided only that the code time frames are time aligned with each other. In an exemplary embodiment, signals are communicated between a cell-site and mobile units using direct sequence spread spectrum communication signals. In the cell-to-mobile link, pilot, sync, paging and voice channels are defined. Information communicated on the cell-to-mobile link channels are, in general, encoded, interleaved, bi-phase shift key (BPSK) modulated with orthogonal covering of each BPSK symbol along with quadrature phase shift key (QPSK) spreading of the covered symbols. In the mobile-to-cell link, access and voice channels are defined. Information communicated on the mobile-to-cell link channels are, in general, encoded, interleaved, orthogonal signaling along with QPSK spreading.
摘要:
A data multiplexing network is described which multiplexes a plurality of asynchronous data channels with an asynchronous data stream representing compressed voice signals and/or facsimile signals onto a single synchronous data packet stream. The single synchronous data packet stream is then transmitted by a high speed statistical multiplexer over a composite link to a second site using a modified high-level synchronous data link control protocol with an overlay of a priority statistical multiplexing algorithm. The asynchronous data channels and the compressed voice channel and/or facsimile signals are demultiplexed and reconstructed for sending to other asynchronous computer terminals and to a standard telephone or facsimile analog port at the second site, respectively. PBX trunk interfaces are also provided to allow PBX's to share the composite link between sites. Communication between the first site by voice or facsimile and the second site is transparent to the users. The multiplexer efficiently allocated the bandwidth of the composite link by detecting silence periods in the voice signals and suppressing the sending of the voice information to preserve bandwidth. The transmission of facsimile information is performed by decoding the fax transmissions, sending only the facsimile image packets over the multiplexed composite link and reassembling, with carrier, the facsimile signals at the remote site for transmission over an analog telephone line to a remote facsimile machine.
摘要:
A splicer for splicing nulllivenull bit streams such as those which carry video programs that have been encoded according to the MPEG-2 standard. The splicer controls the rate at which it outputs the spliced bit stream by means of a model of the receiver and can thereby prevent overflow or underflow in receivers receiving the spliced bit stream. The splicer also includes analyzers for reading the old bit stream and the new bit stream that is to be spliced to the new bit stream. The analyzers provide information to the receiver model and also permit the splicer to select IN and OUT points in the old and new bit streams that minimize the effect of the splice on the decoding of the bit stream done in the receiver. Where necessary, the splicer modifies the output bit stream to reduce interference with decoding. The splicer does not require splice parameters to select IN and OUT points or to determine the proper bit rate or the spliced bit stream. The splicer is further able to make non-seamless and seamless splices and greatly simplifies the making of undetectable splices. It is also able to splice in response to an external splice signal, to a splice command in a bit stream, or to the presence of the beginning or end of a bit stream in the splicer.
摘要:
Method of processing a transmitted digital media data stream. A subsequent data element that follows an unreceived data element in the data stream is received. A parameter of the unreceived data element is estimated based on the received subsequent data element. In one embodiment, each received data element is held in a buffer until a prescribed playout deadline, at which time the data element is released for playout. A loss rate at which data elements in the data stream are not received by their respective playout deadlines is monitored. A time interval extending from the time a data element is sent by a transmitting end to the playout deadline is adjusted based upon the loss rate.
摘要:
A synchronous serial protocol is used to transfer data from a host processor (102) to multiple CODECs (112) operating at differing speeds and from such CODECs (112) back to the host processor (102) across a bidirectional pair of serial links (134, 136). A link speed fast enough to accommodate all operating CODECs (112) is utilized. Frames are transmitted across the bidirectional pair of serial links (134, 136) controlled by frame sync (132) and bit clock (130) signals. Each frame has a control word of validity bits followed by one data word for each active CODEC (112). The validity bits determine whether the corresponding data word contains valid data for or from the corresponding CODEC (112).
摘要:
A system for dynamically allocating a resource is disclosed which includes a plurality of resource users and a resource having a maximum utilization level sharable among the plurality of resource users. A plurality of need analyzers, associated with respective resource users, dynamically generate respective signals (COMPLEXITY), each representing the relative need for the resource by the associated resource user. A plurality of access controllers, associated with respective resource users, control access to the resource by the associated user in response to an allocation signal (CONTROL). A resource allocator dynamically generates allocation signals (CONTROL), representing allocated resource utilization levels for associated users, in response to the plurality of need representative signals (COMPLEXITY) from the need analyzers.
摘要:
A method and apparatus for dynamically allocating the available bandwidth of a common transmission channel of a multiplexed system among multiple encoders in such a manner as to maximize and equalize the quality of the encoded data output by all of the encoders, while also preventing underflow or overflow of encoder or decoder buffers at each end of the common transmission channel, and moreover, while also ensuring compliance with (i.e., without violating) the data encoding and transmission protocol utilized by the system. Further, the bandwidth of the common transmission channel is preferably allocated using an algorithm that does not impose any constraints on the size of the encoder or decoder buffers, other than any constraints specified by the data transmission protocol employed in transmitting the encoded data over the common transmission channel. In a presently preferred embodiment, in which video signals from multiple sources are encoded, both the output channel rate ("bit rate") of each of the encoders and the target number of bits for each picture that is encoded by each encoder are controlled by a controller in accordance with a control algorithm which ensures that the quality of the encoded pictures output by all of the encoders is equalized and maximized, that no underflow or overflow of the encoder or decoder buffers occurs, and that the data encoding and transmission protocol utilized by the system is not violated. Further, the control algorithm employed by the controller preferably does not impose any constraints on the sizes or relative sizes of the encoder or decoder buffers. Moreover, the control algorithm employed by the controller preferably maintains all of the encoder buffers as empty as possible in order to provide increased flexibility to the rate allocation procedure.