During operation radio frames are divided into a plurality of subframes. Data is transmitted over the radio frames within a plurality of subframes, and having a frame duration selected from two or more possible frame durations.

Method and apparatus for reducing round-trip latency and overhead within a communication system

An approach is provided for supporting multi-media conferencing. Textual information (e.g., text associated with a presentation) is received for display during a conference session to multiple participants. For a particular participant, configuration information corresponding to the participant specifies language assistance for the textual information. Language assistance involves augmenting the textual information according to the configuration information for comprehension of the textual information by the particular participant. In one embodiment, the augmented textual information includes language translation of the textual information. Additionally, the augmented textual information can encompass definitions of terms, including acronyms. The augmented textual information is forwarded for display to the participant during the conference session.

Method and system for providing conferencing services

The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Method and apparatus for performing packet loss or frame erasure concealment

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

An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source. In the header reduction technique, the audio encoder selectively modifies the quantization step size of zeroed quantization bands so as to encode in fewer frame header bits.

Quality improvement techniques in an audio encoder

An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such "interaction," for instance, may take the form of updating states in one model using the other model.

Forward error correction in speech coding

Perceptually relevant non-speech information can be preserved during encoding of an audio signal by determining whether the audio signal includes such information (122, 124, 125). If so, a speech/noise classification of the audio signal is overriden (43) to prevent misclassification of the audio signal as noise.

Complex signal activity detection for improved speech/noise classification of an audio signal

A method and system for measuring the speech quality in a mobile cellular telecommunications network using available radio link parameters is disclosed herein. In a preferred embodiment, the method includes receiving a set of radio link parameters, as defined in a standard or otherwise available, such as the BER, FER, RxLev, handover statistics, soft information, and speech energy. Temporal information is obtained from the radio link parameters to create a set of temporal parameters which can be statistically analyzed, for example, for the maximum and minimum, mean, standard deviation, and autocorrelation values for a time interval. The temporal parameters are combined to yield a set of correlated parameters that are more closely related to the speech quality. An estimator then uses the correlated parameters to calculate an estimate for the speech quality. The method of the present invention takes advantage of temporal information and correlated relationships from the transmitted parameters. Furthermore, the method is inherently simple and reliable as compared to prior art methods. Still further, the technique provides a method and allows operator to monitor quality conditions throughout the network.

Speech quality measurement in mobile telecommunication networks based on radio link parameters

An improved method and system of measuring the perceived speech quality in mobile telecommunications networks is disclosed herein. In an embodiment of the invention, the method uses both radio link parameters and an objective measuring technique performed on received signals to estimate the speech quality perceived by the end-user. A radio link processing stage extracts temporal information from a set of available radio link parameters such as the BER, FER, RxLev, handover statistics, soft information, and speech energy. Concurrently, a speech processing stage is used to process a sequence of original signals and received signals, obtained from the output of a telecommunications system. The signal sequences are processed by an objective measuring technique such as Perceptual Speech Quality Measure (PSQM). The outputs from the radio link processing and speech processing stages are utilized to calculate an estimate for speech quality. Furthermore, a weight may be given to radio link processing and speech processing in accordance with their performance under various conditions such that the overall speech quality is calculated with respect to the best approach.

Speech quality measurement based on radio link parameters and objective measurement of received speech signals

A multi-codebook fixed bitrate CELP signal block encoder/decoder includes a codebook selector (22) for selecting, for each signal block, a corresponding codebook identification in accordance with a deterministic selection procedure that is independent of signal type Included are also means for encoding/decoding each signal block by using a codebook having the selected codebook identification

Anders Uvliden

Papers

Forward error correction in speech coding

Complex signal activity detection for improved speech/noise classification of an audio signal

Speech quality measurement in mobile telecommunication networks based on radio link parameters

Speech quality measurement based on radio link parameters and objective measurement of received speech signals

CELP encoding/decoding method and apparatus