Abstract:
A mobile station (106) in a code division multiple access (CDMA) communication system (100) implements scanning of pilot channels on a different frequency by reserving a periodic frame on the forward channel to allow the mobile station (106) to change frequencies and scan for other pilot channels. To preserve some aspects of voice quality, both the base station (102) and mobile station (106) voice encode speech at a maximum of half rate and transmit the information as secondary traffic prior to the frame where the mobile station (106) scans the alternate frequency. To maximize the trade-off between voice quality and frequency of the scan, the base station (102) indicates to the mobile station (106) the period between other frequency scans via messaging. To ensure compatibility, this method can be negotiated via known service configuration negotiation techniques.
Abstract:
A speech coder (300) that performs analysis-by-synthesis coding of a signal determines gain parameters for each constituent component of multiple constituent components of a synthetic excitation signal (ex(n)). The speech coder generates a target vector (p(n)) based on an input signal (s(n)). The speech coder further generates multiple constituent components associated with the synthetic excitation signal, wherein one constituent component of the multiple constituent components is based on a shifted version of another constituent component of the multiple constituent components. The speech coder further evaluates an error criteria based on the target vector and the multiple constituent components to determine a gain associated with each constituent component of the multiple constituent components.
Abstract:
To accurately determine rate and voice activity in moderate-to-low signal-to-noise ratios (item 703) (SNRs) to maximize voice quality, system capacity and/or battery life, parameters from a noise suppression system are used as inputs to the rate determination function. Voice metrics are compared to thresholds (item 715) and rates are determined (items 721, 727, 730).
Abstract:
An open-loop delay contour estimator (204) generates delay information during coding of an information signal. The delay contour is adjusted according to an error minimization criterion on a subframe basis, which allows a more precise estimate of the true delay contour. A delay contour reconstruction block (211) uses the delay information in a decoder in reconstructing the information signal.
Abstract:
Forward link communication capacity of a communication system, which is providing from a base station to a number of mobile stations a number of forward communication links (123) corresponding to the plurality of mobile stations, is controlled by receiving gain setting (102) associated with at least one of the forward communication links, comparing the gain setting with a gain threshold (123), and adjusting a first encoding rate of at least one of the forward communication links to a second encoding rate based on the comparing. A gain offset (310) attunes the gain threshold (123) to update the gain threshold (123) according to a latest condition of the communication system. The gain offset (310) further is used to determine blocking of an incoming call to the communication system.
Abstract:
An improved non-linear processor used in echo cancellation eliminates a comfort noise source (214) and instead inputs a control signal (224) directly into a noise suppression system (403). The noise suppression system (403) uses the control signal (224) to inhibit the iterative update of the background noise estimate when the control signal (224) is active, which prevents any residual echo from biasing the noise estimate provided by the noise suppression system (403). Additionally, the control signal (224) is used by a gain calculator (533) within the noise suppression system (403) to attenuate each frequency band to the maximum allowable amount plus the current residual channel signal-to-noise ratio (SNR). Depending on the implementation, the noise suppression system (403) models the background noise of either a user of the PSTN or a user of a mobile station.
Abstract:
A method and apparatus for improving listener differentiation of talkers during a conference call is provided herein. Particularly, during a teleconference a node (101) will extend the bandwidth of received signals (e.g., speech). Each caller within the conference call will then have their voice projected by the node (101) to a particular spot in three-dimensional space.
Abstract:
The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.
Abstract:
A communication device (10) can include a transceiver (12), a transducer or microphone (14) coupled to the transceiver for receiving voice input from a user, and a processor (16) coupled to the transceiver. The processor can be programmed to receive (32) a predetermined type of phone call or an indication of a caller indication from a calling party and warp (38) a voice input from the user of the communication device based on the caller indication. How the predetermined type of phone call is determined can be done in a number of ways including using an optional caller identification module (18). The caller ID module can provide information on whether the caller is recognized or whether the caller ID information is unavailable.