Abstract: Methods of operation and interrupt controllers for generating interrupt signals to a unit, which could enter an active mode and a non-active mode, are disclosed. The interrupt controllers have interrupt logic (204) adapted for receiving requests for interrupt, activity mode logic (202) adapted for receiving information whether the unit is in non-active mode, and delay control logic (203) adapted for delaying the interrupt to the unit when the received information indicates that the unit is in non-active mode.

Abstract: Control of communication in a data communication system of at least two subsystems is presented. Scheduling transfer of data is performed from a transmitting subsystem to a receiving subsystem. The scheduling comprises determining at least one of a plurality of transfer conditions including a level of activity of each subsystem, a point in time when each subsystem is scheduled to be active, a time limit for receiving data, in the receiving subsystem, an amount of data the receiving subsystem need, and a maximum amount of outstanding data in transfer between said subsystem. In dependence on at least the determined transferring conditions is transferring of data from the transmitting subsystem to the receiving subsystem, the transfer being subject to a delay that depends on the determined at least one transfer condition.

Abstract: A data-processing unit has a fetching circuitry (20) and execution circuitry (30a, 30b). The data-processing unit has an instruction set comprising a nested-loop instruction. The fetching circuitry is arranged to fetch the nested-loop instruction, and the execution circuitry is arranged to execute the nested-loop instruction. The nested-loop instruction comprises at least one instruction field that is adapted to indicate a number of iterations of an outer loop of the nested loop and one or more operations to be performed by the outer loop. Moreover, the at least one instruction field is further adapted to indicate a number of iterations of an inner loop of the nested loop and one or more operations to be performed by the inner loop. A method for fetching, decoding, and executing the nested-loop instruction is also described as well as the structure of the nested-loop instruction.

Abstract: A data-processing unit comprises a register unit (10) comprising a register (20, R0-R3). The data-processing unit further comprises an address-generation unit (30) for generating a memory address to a memory unit (60). The address-generation unit (30) is adapted to fetch, from the register (20, R0-R3), a base address stored in a first portion (20a) of the register (20, R0-R3) and a first offset address stored in a second portion (20b) of the register (20, R0-R3). The base address and the first offset address are represented with fewer bits than the memory address. The address generation unit (30) is adapted to receive a first instruction and, in response thereto, generate a second offset address based on the first offset address, and generate the memory address by adding the base address and the second offset address. A method for generating the memory address is also disclosed.

Abstract: A processor for processing data is provided. The processor comprises an address generator, which is operative to generate an address based on a base address and a fractional step (?).

Abstract: Realistic simulation of acoustic obstruction/occlusion effects in virtual-reality software applications is achieved by specifying whether a type of filter function is low-pass or high-pass and a cut-off frequency and stop-band attenuation of the filter function. The stop-band attenuation can be specified merely qualitatively, for example as “weak”, “nominal”, or “strong”. As a complement or alternative, obstruction/occlusion can be specified in terms of obstruction objects, such as blocking objects, enclosure objects, surface objects, and medium objects. An obstruction object is specified in terms of one or more environmental parameters and corresponds to naturally occurring acoustically obstructive/occlusive objects, such as curtains, walls, forests, fields, etc. The two specification types—filter specification parameters and environmental parameters—may co-exist in the same implementation or one or the other of the interfaces can be used in a particular implementation.

Abstract: A method and unit for composing or decomposing a multimedia signal according to the Musical Instrument Digital Interface (MIDI) protocol. The signal carries events of a first type, which are arranged to carry instructions regarding which predefined patches to use for playback and which predefined notes to play, and events of a second type which carry additional content. The signal is parsed to identify events of the second type and to read the additional content. Coded samples of multimedia content are loaded at an address specified in the additional content. The coded samples are then decoded to provide the samples for playback of the multimedia content. In this manner, vocal song or vocals and other audio type signals are efficiently conveyed utilizing the widely used MIDI protocol.

Abstract: A common narrow-band speech signal is expanded into a wide-band speech signal. The expanded speech signal gives the impression of a wide-band speech signal regardless of what type of vocoder is used. Extending the narrow-band speech signal into a lower range involves analyzing the narrow-band speech signal to generate one or more parameters, and synthesizing a lower frequency-band signal based on at least one of the one or more parameters. The synthesized lower frequency-band signal is then combined with a signal that is derived from (e.g., via up-sampling) the narrow-band speech signal. In preferred embodiments, a pitch frequency parameter is generated, and generation of the lower frequency-band signal includes generating continuous sine tones that are frequency shifted with the pitch frequency parameter.

Abstract: Speech enhancement is provided in dual microphone noise reduction systems by including spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, when a far-mouth microphone is used in conjunction with a near-mouth microphone, it is possible to handle non-stationary background noise as long as the noise spectrum can continuously be estimated from a single block of input samples. The far-mouth microphone, in addition to picking up the background noise, also picks up the speaker's voice, albeit at a lower level than the near-mouth microphone. To enhance the noise estimate, a spectral subtraction stage is used to suppress the speech in the far-mouth microphone signal. To be able to enhance the noise estimate, a rough speech estimate is formed with another spectral subtraction stage from the near-mouth signal.

Abstract: A system and method for speech signal enhancement upsamples a narrowband speech signal at a receiver to generate a wideband speech signal. The lower frequency range of the wideband speech signal is reproduced using the received narrowband speech signal. The received narrowband speech signal is analyzed to determine its formants and pitch information. The upper frequency range of the wideband speech signal is synthesized using information derived from the received narrowband speech signal.

Abstract: Speech enhancement is provided in dual microphone noise reduction systems by including spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, when a far-mouth microphone is used in conjunction with a near-mouth microphone, it is possible to handle non-stationary background noise as long as the noise spectrum can continuously be estimated from a single block of input samples. The far-mouth microphone, in addition to picking up the background noise, also picks up the speaker's voice, albeit at a lower level than the near-mouth microphone. To enhance the noise estimate, a spectral subtraction stage is used to suppress the speech in the far-mouth microphone signal. To be able to enhance the noise estimate, a rough speech estimate is formed with another spectral subtraction stage from the near-mouth signal.

Abstract: For purposes of noise suppression, spectral subtraction filtering is performed in sample-wise fashion in the time domain using a time-domain representation of a spectral subtraction gain function computed in block-wise fashion in the frequency domain. By continuously performing time-domain filtering on a sample by sample basis, the disclosed methods and apparatus avoid block-processing delays associated with frequency-domain based spectral subtraction systems. Consequently, the disclosed methods and apparatus are particularly well suited for applications requiring very short processing delays. Moreover, since the spectral subtraction gain function is computed in a block-wise fashion in the frequency domain, high quality performance in terms of reduced tonal artifacts and low signal distortion is retained.

Abstract: For purposes of noise suppression, spectral subtraction filtering is performed in sample-wise fashion in the time domain using a time-domain representation of a spectral subtraction gain function computed in block-wise fashion in the frequency domain. By continuously performing time-domain filtering on a sample by sample basis, the disclosed methods and apparatus avoid block-processing delays associated with frequency-domain based spectral subtraction systems. Consequently, the disclosed methods and apparatus are particularly well suited for applications requiring very short processing delays. In applications where only stationary, low-energy background noise is present, computational complexity is reduced by generating a number of separate spectral subtraction gain functions during an initialization period, each gain function being suitable for one of several predefined classes of input signal (e.g.

Abstract: Methods and apparatus for providing speech enhancement in noise reduction systems include spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, successive blocks of a spectral subtraction gain function are averaged based on a discrepancy between an estimate of a spectral density of a noisy speech signal and an averaged estimate of a spectral density of a noise component of the noisy speech signal. The successive gain function blocks are averaged, for example, using controlled exponential averaging. Control is provided, for example, by making a memory of the exponential averaging inversely proportional to the discrepancy. Alternatively, the averaging memory can be made to increase in direct proportion with decreases in the discrepancy, while exponentially decaying with increases in the discrepancy to prevent audible voice shadows.

Abstract: A common narrow-band speech signal is expanded into a wide-band speech signal. The expanded speech signal gives the impression of a wide-band speech signal regardless of what type of vocoder is used. Extending the narrow-band speech signal into a lower range involves analyzing the narrow-band speech signal to generate one or more parameters, and synthesizing a lower frequency-band signal based on at least one of the one or more parameters. The synthesized lower frequency-band signal is then combined with a signal that is derived from (e.g., via up-sampling) the narrow-band speech signal. In preferred embodiments, a pitch frequency parameter is generated, and generation of the lower frequency-band signal includes generating continuous sine tones that are frequency shifted with the pitch frequency parameter.

Abstract: A common narrow-band speech signal is expanded into a wide-band speech signal. The expanded signal gives the impression of a wide-band speech signal regardless of what type of vocoder is used in a receiver. The robust techniques suggested herein are based on speech acoustics and fundamentals of human hearing. That is the techniques extend the harmonic structure of the speech signal during voiced speech segments and introduce a linearly estimated amount of speech energy in the wide frequency-band. During unvoiced speech segments, a fricated noise may be introduced in the upper frequency-band.

Abstract: A method of transmitting voice information from an electronic communications device (1) comprises the steps of receiving the voice information from the environment of the device together with a first background sound, generating a sound signal having a first signal part representing the voice information and a second signal part representing the first background sound, reducing the signal part representing the first background sound, and transmitting the sound signal through a communications channel to which the device is connected. The method further comprises the step of adding to the sound signal an additional signal representing a second background sound. In this way background noise can be removed, while a natural and comfortable conversation can be maintained without revealing the location of the user of the device.

Abstract: A system and method for speech signal enhancement upsamples a narrowband speech signal at a receiver to generate a wideband speech signal. The lower frequency range of the wideband speech signal is reproduced using the received narrowband speech signal. The received narrowband speech signal is analyzed to determine its formants and pitch information. The upper frequency range of the wideband speech signal is synthesized using information derived from the received narrowband speech signal.

Abstract: Speech enhancement is provided in dual microphone noise reduction systems by including spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, when a far-mouth microphone is used in conjunction with a near-mouth microphone, it is possible to handle non-stationary background noise as long as the noise spectrum can continuously be estimated from a single block of input samples. The far-mouth microphone, in addition to picking up the background noise, also picks up the speaker's voice, albeit at a lower level than the near-mouth microphone. To enhance the noise estimate, a spectral subtraction stage is used to suppress the speech in the far-mouth microphone signal. To be able to enhance the noise estimate, a rough speech estimate is formed with another spectral subtraction stage from the near-mouth signal.

Abstract: Methods and apparatus for providing speech enhancement in noise reduction systems include spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, low order spectrum estimates are developed which have less frequency resolution and reduced variance as compared to spectrum estimates in conventional spectral subtraction systems. The low order spectra are used to form a gain function having a desired low variance which in turn reduces musical tones in the spectral subtraction output signal. Advantageously, the gain function can be further smoothed across blocks using input spectrum dependent exponential averaging. Additionally, the low order of the gain function permits a phase to be added during interpolation so that the spectral subtraction gain filter is causal and prevents discontinuities between blocks.