Samsung Electronics Co., Ltd.

Nokia Corporation

Other

Nokia Technologies Oy

Nytell Software LLC

Steven Craig Greer

Hosang Sung

Zhigang Rong

Lin L Ma

Hussein S El-Ghoroury

Milan Jelinek

Philippe Gournay

Zhigang Liu

Zhouyue Pi

Bo Wei

An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec, configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or mom FEC modes. Upon the setting of the operation mode to be the High FER mode, the one FEC mode is selected, from the one or more FED modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Frame erasure concealment for a multi-rate speech and audio codec

An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec, configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode, the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Frame erasure concealment for a multi rate speech and audio codec

A method, apparatus and system for scheduling data packets in a wireless communications system. The invention includes determining a priority for at least one data packet, wherein determining the priority of the data packet is based at least on a plurality of quality of service factors, wherein each of the plurality of QoS factors has a corresponding weighting factor and the determined priority includes minimum-performance guarantees. The invention further includes scheduling transmission of the at least one data packet based at least on the determined priority.

Method and apparatus to facilitate real-time packet scheduling in a wireless communications system

The present invention concerns methods for efficiently supporting Voice over Internet Protocol (VoIP) on the Forward Packet Data Channel (F-PDCH) in CDMA 2000 1xEV-DV systems. Active speech in VoIP is encoded using, for example enhanced variable rate codec (EVRC), which produces 171, 80 and 16 bits per 20 ms of speech for Rate 1, Rate &#xbd; and Rate &#x215b;, respectively. However, about 60% of the time a user is inactive during a speech session, so an inordinate amount of system bandwidth is comprised of rate &#x215b; VoIP packets. In one embodiment of the present invention the apparatus of the present invention identifies the Rate &#x215b; voice frame packets and discards them. In another embodiment of the present invention, the apparatus of the present invention identifies the Rate &#x215b; voice frame packets and selects some of them for further transmission. In both embodiments the efficiency of channel utilization is increased since the amount of channel band width used to communicate relatively little information, e.g., gaps of silence, is decreased.

Apparatus and method for efficiently supporting VoIP in a wireless communication system

Apparatus is provided that includes at least one entity for transmitting speech signals in a discontinuous transmission mode including transmitting speech frames interspersed with frames including comfort noise parameters during periods of speech pauses. The entit(ies) include a first entity for estimating a current noise value. In addition, the apparatus includes a second entity for selectively controlling a rate at which the frames including comfort noise parameters are transmitted during the periods of speech pauses based upon the estimated current noise value.

System and method for adaptive transmission of comfort noise parameters during discontinuous speech transmission

The use of a cellular system's packet data channel for transport of speech communications is enhanced by dynamically controlling the transmission of data packets that were received in error and notified to the transmitter by the receiver in accordance with an algorithm that takes into account a self-generated determination of the perceptual quality importance of the frame and is generated once for each frame by the speech encoder. The calculation of the frame importance is codec-dependent, and depends on a number of attributes associated with each particular frame, including the frame rate (for variable-rate codecs), signal classification, transition characteristics, background noise, previous frame erasures, etc. The selective reduction in transmission and retransmission increases the capacity of the cellular system while maintaining or even improving the quality of the received voice signal.

Codec-assisted capacity enhancement of wireless VoIP

A system for self-calibrating a clock of a communication terminal for use with communication systems in which a central communication node generates time base correction signals for the terminal clock includes a terminal oscillator which generates an oscillator frequency that includes an error amount. An oscillator calibration filter generates a frequency error estimate amount. Circuitry is provided for subtracting the frequency error estimate amount generated by the calibration filter from the oscillator frequency error amount. Circuitry is provided for applying the time base correction signals to the calibration filter to thereby modify the frequency error estimate amount generated by the calibration filter based upon the time base correction signals generated by the communication central node.

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Rowlett, TX, United States of America

Steven Craig Greer