Abstract: An overflow problem of LSF quantization in G.729 Annex B speech encoding which may lead to non-assignment of a codebook index. Preferred embodiments fix the problem with default or limited random variable assignments or flagging the overflow and adjusting the frame encoding such as by limiting spectral components or changing quantization targets.

Abstract: A method to reduce memory requirements for a packet loss concealment algorithm in the event of packet loss in a receiver of pulse code modulated voice signals. Packet losses are concealed by using the spectral analysis filter memory to smooth a signal gap and by using a technique for determining a maximum repeatable waveform range instead of using the pitch period to reproduce lost packets. The invention uses fewer processing resources and results in improved performance compared to a packet loss concealment algorithm under G.711 Appendix I standards.

Abstract: Embodiments of the invention provide methods to calculate compression noise statistics of decompressed images in transform coding. They can be used in compressed image quality assessment, compression algorithm optimization, compression noise reduction, and other quantization and compression related applications.

Abstract: In a voice activity detection (VAD) device a method for defining tone signals comprises defining a threshold for zero amplitude change, calculating a zero crossing rate of a signal, extracting a set of parameters from a plurality of duration periods of the signal, defining a tolerance threshold between the plurality of duration periods when a zero amplitude change occurs, calculating a maximum difference between the plurality of duration periods, and comparing the maximum difference with the threshold. The method is implemented in the International Telecommunications Union (ITU) recommendation G.729 Annex B VAD.

Abstract: A method to improve packet loss concealment for generation of a synthetic speech signal in a algebraic code excited linear prediction decoder for a voice over packet network. One method improves features for coding gains in the decoder and for post-filtering of the signals. An alternative method uses a classification method for the signal based on the bitstream in the decoder.

Abstract: ITU Recommendation G.729 Annex E teaches in the implementation of a fixed codebook search to determine the selected sample combination providing the minimal difference between the original input speech and the reconstructed speech after implementation of the codec. A large number of sample sets are processed and the difference between the original input signal and the reconstructed signal for each set is determined and stored in a register. Under certain conditions, the register can overflow resulting in invalid difference values. When such a condition occurs, the fixed codebook search cannot determine the sample combination providing the minimal mean square error between the weighted input speech and the weighted reconstructed speech. An initialization vector for the codvec vector is used to provide valid data which conforms to the G.729 Annex E specifications and minimizes changes to the G.729 source code while providing robust quality signal processing in the event of register overflow condition.

Abstract: A distributed intelligence conferencing system having a plurality of conferencing nodes to connect groups of participants to a conference. Each of the conferencing nodes provides for the connection of one or more participants to the conference. Each node includes a DSP for distributed signal processing. The nodes DSP includes: A signal measuring device for measuring features of the signals from each of the participants. A single core speaker tracking algorithm identifies the active speaker on a local node based upon the relative signal features of each of the number of participant input signals. A communication device communicates the measured signal features for the active local participant input signals only to all other conferencing nodes. A multi core speaker tracking algorithm compares the features of the signals from the speakers on the local node with the features of the signals from the speakers on other nodes.

Abstract: In a voice activity detection (VAD) device a method for defining tone signals comprises defining a threshold for zero amplitude change, calculating a zero crossing rate of a signal, extracting a set of parameters from a plurality of duration periods of the signal, defining a tolerance threshold between the plurality of duration periods when a zero amplitude change occurs, calculating a maximum difference between the plurality of duration periods, and comparing the maximum difference with the threshold. The method is implemented in the International Telecommunications Union (ITU) recommendation G.729 Annex B VAD.

Abstract: A distributed conferencing system having a plurality of conferencing nodes to connect groups of participants to a conference. Each of the conferencing nodes provides for the connection of one or more participants to the conference. Each node includes a DSP for distributed signal processing. The node DSP includes: A signal measuring device for measuring features of the signals from each of the participants, such as power, zero crossing rate and short term energy and voice activity determination and feature extraction from the signals of the participants connected to each node. Each node has a single core speaker tracking algorithm for determining the relative features of each of the number of participant input signals. The speaker tracking compares the characteristics of the speakers of the core and determines which speakers are to be included and which speakers are to be excluded from presentation to the other nodes in the conference.

Abstract: A technique is described to change the codec or MAC (message authentication code) size in a packet security unit for PacketCable communications during realtime voice transmissions is described. An algorithm that provides fast RC4 key advancing to prevent MIPS (millions of instruction cycles per second) overflow is used to perform codec or MAC size changes. The invention is performed without changing the keying material, where the sender and receiver must continue the RC4 encryption process from its state prior to the codec or MAC size change. A sender needs to preserve continuity of the timestamp across a codec change, since the timestamp reflects realtime. Changing the codec or MAC size is likely to change the frame parameters. To preserve continuity of the RC4 state and the timestamp across the codec/MAC size change, the sender TX and receiver RX generates a new frame number. The new frame number is applied to the first frame generated by the new codec or MAC size.

Abstract: A method to reduce memory requirements for a packet loss concealment algorithm in the event of packet loss in a receiver of pulse code modulated voice signals. A voice playout unit in the receiver shares its nominal delay buffer with a history buffer of a packet loss concealment algorithm up to a maximum limit described in a standard. This reduces or eliminates need to allocate memory for the history buffer. A history buffer can also be extended to retain an original portion of voice signal packets received prior to a packet loss as well as generated voice signals as they are generated. A scratch buffer is used as a working buffer and replaces the function of a pitch buffer.

Abstract: A method to reduce memory requirements for a packet loss concealment algorithm in the event of packet loss in a receiver of pulse code modulated voice signals. Packet losses are concealed by using the spectral analysis filter memory to smooth a signal gap and by using a technique for determining a maximum repeatable waveform range instead of using the pitch period to reproduce lost packets. The invention uses fewer processing resources and results in improved performance compared to a packet loss concealment algorithm under G.711 Appendix I standards.

Abstract: In a packet-based multimedia communication system, such as ITU G.711, using linear prediction parameters to derive the linear prediction error of the codec. The linear prediction error is then used as a feature of the Silence Insertion Descriptor (SID) algorithm. Generating a SID frame by comparing linear prediction errors between frames in the input data stream to a threshold.

Abstract: A method of initializing an ITU Recommendation G.729 Annex B compliant voice activity detection (VAD) device is disclosed, having the steps of (1) determining a first set of running average background noise characteristics in accordance with Recommendation G.729B; (2) determining a second set of running average background noise characteristics; and (3) substituting the second set of running average background noise characteristics for the first set when a specific event occurs. The specific event is a divergence between the first and second sets of running average background noise characteristics.

Abstract: A method of initializing an ITU Recommendation G.729 Annex B voice activity detection (VAD) device is disclosed, having the steps of (1) extracting a set of parameters from a signal that characterize the signal; (2) calculating an energy measure of the signal from the set of parameters; (3) comparing the energy measure with a reference value; (4) determining an initial value for an average of a noise characteristic of the signal; and (5) counting the number of times the energy measure equals or exceeds the reference level. Also disclosed is a method of converging an ITU Recommendation G.

Abstract: A distributed conferencing system has a plurality of conferencing nodes to connect groups of participants to a conference. Each of the conferencing nodes provides for the connection of one or more participants to the conference. Each node includes a DSP for distributed signal processing. The node DSP includes: A signal measuring device for measuring features of the signals from each of the participants such as power, zero crossing rate and short term energy. The nodes include voice activity determination and a communication device for communicating the measured signal characteristics for a plurality of participant input signals to all other conferencing nodes. Muting means for muting individual participant input signals so that only selected signals are transmitted over the conference bus to the other participants.

Abstract: In a voice activity detection (VAD) device a method for defining tone signals comprises defining a threshold for zero amplitude change, calculating a zero crossing rate of a signal, extracting a set of parameters from a plurality of duration periods of the signal, defining a tolerance threshold between the plurality of duration periods when a zero amplitude change occurs, calculating a maximum difference between the plurality of duration periods, and comparing the maximum difference with the threshold. The method is implemented in the International Telecommunications Union (ITU) recommendation G.729 Annex B VAD.

Abstract: A method to reduce the amount of bandwidth used in the transmission of digitized voice packets is described. The method is used to reduce the number of transmitted packets by suspending transmission during periods of silence or when only noise is present. The system determines if a background noise update is warranted based on human auditory perception factors instead of an artificial limiter on excessive silence insertion descriptor packets. The system searches for characteristics in the perceptual changes of background noise instead of analyzing speech for improved audio compression. The invention weighs factors affecting the perception of sound including frequency masking, temporal masking, loudness perception based on tone, and auditory perception differential based on tone.

Abstract: A technique is described to change the codec or MAC (message authentication code) size in a packet security unit for PacketCable communications during realtime voice transmissions is described. An algorithm that provides fast RC4 key advancing to prevent MIPS (millions of instruction cycles per second) overflow is used to perform codec or MAC size changes. The invention is performed without changing the keying material, where the sender and receiver must continue the RC4 encryption process from its state prior to the codec or MAC size change. A sender needs to preserve continuity of the timestamp across a codec change, since the timestamp reflects realtime. Changing the codec or MAC size is likely to change the frame parameters. To preserve continuity of the RC4 state and the timestamp across the codec/MAC size change, the sender TX and receiver RX generates a new frame number. The new frame number is applied to the first frame generated by the new codec or MAC size.

Abstract: ITU Recommendation G.729 Annex E teaches in the implementation of a fixed codebook search to determine the selected sample combination providing the minimal difference between the original input speech and the reconstructed speech after implementation of the codec. A large number of sample sets are processed and the difference between the original input signal and the reconstructed signal for each set is determined and stored in a register. Under certain conditions, the register can overflow resulting in invalid difference values. When such a condition occurs, the fixed codebook search cannot determine the sample combination providing the minimal mean square error between the weighted input speech and the weighted reconstructed speech. An initialization vector for the codvec vector is used to provide valid data which conforms to the G.729 Annex E specifications and minimizes changes to the G.729 source code while providing robust quality signal processing in the event of register overflow condition.