Abstract: A method and apparatus for performing a active packet bundling in a VoIP (Voice over Internet Protocol) based on a measure of voice concealability of one or more speech packets to be transmitted. Voice concealability is a property of a speech packet which indicates the likelihood that a given speech segment will meet the “quasi-stationary” (QS) behavior assumed by conventional Packet Loss Concealment (PLC) algorithms used to compensate for transmission loss of speech packets. By ensuring that only well-concealed frames are delayed for bundling, it is more likely that the receiver will run the PLC algorithm when it is most effective (i.e., when it is likely to perform well at reconstructing the missing speech frame), rather than when it is more likely to be ineffective. This allows for a reduction in bandwidth while minimizing any potential negative impact on quality.

Abstract: A video codec is provided in which virtual reference data (e.g., virtual reference blocks) are advantageously employed by a video coder and decoder to significantly improve coding efficiency as compared to prior art codecs. Specifically, a “virtual reference,” as used herein, is defined as a group of pixels (e.g., a block) that is used as reference material for encoding portions of the video signal (e.g., a motion-compensated inter-predicted block), but that does not comprise or represent any portion of the actual video sequence to be displayed. For example, it may advantageously be determined that portions of the video sequence may be efficiently represented as a motion-compensated block of pixel values which are generated by the video coder based on the blocks being predicted but not based on any actual blocks in any of the actual video frames or slices.

Abstract: A method and apparatus for routing voice packets through a communications network wherein the routing priority of the packets is based on a location of a packet within a sequence of transmitted voice packets, relative to one or more identified voice packets in the transmitted sequence. In one embodiment of the invention, the routing priority of a voice packet is based on the location of the packet in a sequence of packets relative to voice activity, such as the beginning, middle or end of a talk spurt. For example, the routing priority of the first several voice packets of a talk spurt is increased to compensate for the fact that there is less network jitter protection available at the receiver for these packets. In another embodiment, the routing priority of a voice packet may depend upon its location relative to a cellular handoff from one transmitter to another.

Abstract: A method and apparatus for enhancing voice intelligibility for network communications of speech such as, for example, VoIP (Voice-Over-Internet-Protocol), in the presence of packets which arrive too late for normal playout. When a late speech packet is received by a speech decoder, that packet and, if necessary, one or more additional packets subsequent thereto, are played out over a shorter than normal duration so that the decoder can “catch up” with the encoder. Since a voice frame is usually decoded in several sub-frames—typically two or three—this shortened playout may be achieved, for example, by skipping one sub-frame from each frame to be shortened.

Abstract: A video codec is provided in which virtual reference data (e.g., virtual reference blocks) are advantageously employed by a video coder and decoder to significantly improve coding efficiency as compared to prior art codecs. Specifically, a “virtual reference,” as used herein, is defined as a group of pixels (e.g., a block) that is used as reference material for encoding portions of the video signal (e.g., a motion-compensated inter-predicted block), but that does not comprise or represent any portion of the actual video sequence to be displayed. For example, it may advantageously be determined that portions of the video sequence may be efficiently represented as a motion-compensated block of pixel values which are generated by the video coder based on the blocks being predicted but not based on any actual blocks in any of the actual video frames or slices.

Abstract: An enhanced Base Transceiver Stations (BTS) includes a satellite receiver for use in receiving video and/or data signals. Since most video services (and many data services) are updated relatively infrequently, and since most such services require that the majority of traffic be sent from the core to the client (as opposed to from the client to the core), delivering such services over a satellite channel advantageously provides an opportunity to offer broadcast services, for example, with minimal use of the existing backhaul network, thereby advantageously eliminating the bottleneck typically created by the limited bandwidth of the backhaul network which is conventionally used to supply such video and/or data signals.

Abstract: A method and apparatus for routing voice packets through a communications network wherein the routing priority of the packets is based on a location of a packet within a sequence of transmitted voice packets, relative to one or more identified voice packets in the transmitted sequence. In one embodiment of the invention, the routing priority of a voice packet is based on the location of the packet in a sequence of packets relative to voice activity, such as the beginning, middle or end of a talk spurt. For example, the routing priority of the first several voice packets of a talk spurt is increased to compensate for the fact that there is less network jitter protection available at the receiver for these packets. In another embodiment, the routing priority of a voice packet may depend upon its location relative to a cellular handoff from one transmitter to another.

Abstract: The present invention is a method for improving data transfer performance over communications networks connecting data networks and users using adaptive communications formatting. Adaptive communications formatting includes encoding (or compressing) the data and applying error control schemes to reduce the amount of data being transmitted and to correct and/or conceal errors occurring during data transmission. In one embodiment, the present invention uses a set of transcoding techniques to encode (or compress) the data and a set of error control schemes to correct and/or conceal errors occurring during data transmission. The particular sets of transcoding techniques and error control schemes selected to format the data are adaptive to factors, such as the nature of the communications network connecting a user to an access server on the data network, the preferences of the user, and the data type of the data being transmitted to the user (or the access server).

Abstract: Automatic Speech Recognition (ASR) is achieved in wireless communications systems in which reliable ASR feature vector sequences are derived at a base station directly from digitally transmitted speech coder parameters, with no additional processing or signal modification required at the originating handset. No secondary channel need be provided for the transmission of ASR feature vectors. In operating on received speech coder parameters prior to conversion to a voice signal the present system and methods avoid the lossy conversion process and associated voice distortion. Since the received voice parameters are error protected during transmission they are received with greater accuracy. All, or a subset, of speech coding parameters, including, in appropriate cases, spectral envelope parameters, reflection coefficients, LSPs, LSFs, LPCs, LPCCs, and weighted LPCCs may be processed at a receiving base station or forwarded to another location for processing.

Abstract: A method and apparatus for improving sound quality in a digital cellular radio system receiver. A voice activity detector uses an energy estimate to detect the presence of speech in a received speech signal in a noise environment. When no speech is present the system attenuates the signal and inserts low pass filtered white noise. In addition, a set of high pass filters are used to filter the signal based upon the background noise level. This high pass filtering is applied to the signal regardless of whether speech is present. Thus, a combination of signal attenuation with insertion of low pass filtered white noise during periods of non-speech, along with high pass filtering of the signal, improves sound quality when decoding speech which has been encoded in a noisy environment.