The present invention relates to a method and device for improving concealment of frame erasure caused by frames of an encoded sound signal erased during transmission from an encoder (106) to a decoder (110), and for accelerating recovery of the decoder after non erased frames of the encoded sound signal have been received. For that purpose, concealment/recovery parameters are determined in the encoder or decoder. When determined in the encoder (106), the concealment/recovery parameters are transmitted to the decoder (110). In the decoder, erasure frame concealment and decoder recovery is conducted in response to the concealment/recovery parameters. The concealment/recovery parameters may be selected from the group consisting of: a signal classification parameter, an energy information parameter and a phase information parameter. The determination of the concealment/recovery parameters comprises classifying the successive frames of the encoded sound signal as unvoiced, unvoiced transition, voiced transition, voiced, or onset, and this classification is determined on the basis of at least a part of the following parameters: a normalized correlation parameter, a spectral tilt parameter, a signal-to-noise ratio parameter, a pitch stability parameter, a relative frame energy parameter, and a zero crossing parameter.

Method and device for efficient frame erasure concealment in linear predictive based speech codecs

Techniques for efficiently assigning resources for spurts of traffic in a wireless communication system are described. The system may support semi-persistent and non-persistent resource assignments. A semi-persistent resource assignment is valid for as long as more data is sent within a predetermined time period of last sent data and expires if no data is sent within the predetermined time period. A non-persistent resource assignment is valid for a predetermined duration or a specific transmission. A semi-persistent resource assignment may be granted for an expected spurt of data to send via the communication link. For Voice-over-Internet Protocol (VoIP), a semi-persistent resource assignment may be granted for a voice frame in anticipation of a spurt of voice activity, and a non-persistent resource assignment may be granted for a silence descriptor (SID) frame during a period of silence.

Semi-persistent scheduling for traffic spurts in wireless communication

A method for including an identifier with a packet associated with a speech signal is described. A signal is received. The signal is partitioned into a plurality of frames. A frame of the signal is encoded into a packet. A determination is made if the packet is encoded as a wideband packet or a narrowband packet. An identifier is packed in the packet based on the determination. The packet is transmitted. At least two illegal values are provided from an N-bit parameter, wherein at least one bit from the N-bit parameter is used to carry information. A number of bits from the N-bit parameter that are used to carry information is equal to log2(X), wherein X is the number of illegal values provided from the N-bit parameter.

Systems and methods for including an identifier with a packet associated with a speech signal

Applications of dim-and-burst techniques to coding of wideband speech signals are described. Reconstruction of a highband portion of a frame of a wideband speech signal using information from a previous frame is also described.

Systems, methods, and apparatus for wideband encoding and decoding of active frames

Techniques for generalized pilot interference cancellation in a communications receiver. In an exemplary embodiment, a residual pilot is cancelled from a post-traffic cancellation signal following initial first-pass pilot cancellation. Residual pilot cancellation is achieved by adding the first-pass cancelled pilot as earlier stored in memory back to the post-traffic cancellation signal, and pilot filtering the resulting signal to generate an improved pilot interference estimate. In an alternative exemplary embodiment, an arbitrary number of iterations may be applied to generate the pilot interference estimate by successively storing each generated pilot interference estimate in memory.

Pilot interference cancellation

A source-controlled Variable bit-rate Multi-mode WideBand (VMR-WB) speech codec, having a mode of operation that is interoperable with the Adaptive Multi-Rate wideband (AMR-WB) codec, the codec comprising: at least one Interoperable full-rate (1-FR) mode, having a first bit allocation structure based an one of a AMR-WB codec coding types; and at least one comfort noise generator (CNG) coding type for encoding inactive speech frame having a second bit allocation structure based on AMR-WB SIDlUPDATE coding type. Methods for i) digitally encoding a sound using a source-controlled Variable bit rate multi-mode wideband (VMR-WB) speech codec for interoperation with an adaptative multi-rate wideband (AMR-WB) codec, ii) translating a Variable bit rate multi-mode wideband (VMR-WB) speech codec-signal frame into an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame, iii) translating an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame into a Variable bit rate multi-mode wideband (VMR-WB) speech signal frame, and iv) translating an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame into a Variable bit rate multi-mode wideband (VMR-WB) speech signal frame are also provided.

Methods for interoperation between adaptive multi-rate wideband (amr-wb) and multi-mode variable bit-rate wideband (wmr-wb) speech codecs

Speech signal classification and encoding systems and methods are disclosed herein. The signal classification is done in three steps each of them discriminating a specific signal class. First, a voice activity detector (VAD) discriminates between active and inactive speech frames. If an inactive speech frame is detected (background noise signal) then the classification chain ends and the frame is encoded with comfort noise generation (CNG). If an active speech frame is detected, the frame is subjected to a second classifier dedicated to discriminate unvoiced frames. If the classifier classifies the frame as unvoiced speech signal, the classification chain ends, and the frame is encoded using a coding method optimized for unvoiced signals. Otherwise, the speech frame is passed through to the 'stable voiced' classification module. If the frame is classified as stable voiced frame, then the frame is encoded using a coding method optimized for stable voiced signals. Otherwise, the frame is likely to contain a non-stationary speech segment such as a voiced onset or rapidly evolving voiced speech signal. In this case a general-purpose speech coder is used at a high bit rate for sustaining good subjective quality .

Methods and devices for source controlled variable bit-rate wideband speech coding

For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame. For searching pitch pulses in a sound signal, a residual signal is produced by filtering the sound signal through a linear prediction analysis filter, a weighted sound signal is produced by processing the sound signal through a weighting filter, the weighted sound signal being indicative of signal periodicity, a synthesized weighted sound signal is produced by filtering a synthesized speech signal produced during a last subframe of a previous frame of the sound signal through the weighting filter, a last pitch pulse of the sound signal of the previous frame is located from the residual signal, a pitch pulse prototype of given length is extracted around the position of the last pitch pulse of the sound signal of the previous frame using the synthesized weighted sound signal, and the pitch pulses are located in a current frame using the pitch pulse prototype.

Signal modification method for efficient coding of speech signals

The present invention relates to a method and device for quantizing linear prediction parameters in variable bit-rate sound signal coding, in which an input linear prediction parameter vector is received, a sound signal frame corresponding to the input linear prediction parameter vector is classified, a prediction vector is computed, the computed prediction vector is removed from the input linear prediction parameter vector to produce a prediction error vector, and the prediction error vector is quantized. Computation of the prediction vector comprises selecting one of a plurality of prediction schemes in relation to the classification of the sound signal frame, and processing the prediction error vector through the selected prediction scheme. The present invention further relates to a method and device for dequantizing linear prediction parameters in variable bit-rate sound signal decoding, in which at least one quantization index and information about classification of a sound signal frame corresponding to the quantization index are received, a prediction error vector is recovered by applying the index to at least one quantization table, a prediction vector is reconstructed, and a linear prediction parameter vector is produced in response to the recovered prediction error vector and the reconstructed prediction vector. Reconstruction of the prediction vector comprises processing the recovered prediction error vector through one of a plurality of prediction schemes depending on the frame classification information.

Method and device for robust predictive vector quantization of linear prediction parameters in variable bit rate speech coding

In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters. When classification of the sound signal determines that the signal-coding parameters should be transmitted using the first communication mode and when a request to transmit the signal-coding parameters from the coder of one station to the decoder of the other station using a second communication mode designed to reduce bit rate during transmission of the signal-coding parameters is received, a portion of the signal-coding parameters from the coder one station is dropped and the remaining signal-coding parameters are transmitting to the decoder of the other station using the second communication mode. The dropped portion of the signal-coding parameters are regenerated before the decoder of the other station decodes the signal-coding parameters.

Method and device for efficient in-band dim-and-burst signaling and half-rate max operation in variable bit-rate wideband speech coding for cdma wireless systems

Milan Jelinek

Papers

Methods for interoperation between adaptive multi-rate wideband (amr-wb) and multi-mode variable bit-rate wideband (wmr-wb) speech codecs

Methods and devices for source controlled variable bit-rate wideband speech coding

Signal modification method for efficient coding of speech signals

Method and device for robust predictive vector quantization of linear prediction parameters in variable bit rate speech coding

Method and device for efficient in-band dim-and-burst signaling and half-rate max operation in variable bit-rate wideband speech coding for cdma wireless systems