Abstract: An apparatus (213) and corresponding methods (FIG. 7) to facilitate maintaining crypto synchronization while processing communication signals in a communication unit includes a vocoder (215) configured to convert input audio band signals to vocoder output frames; a crypto processor (217) configured to encrypt the vocoder output frames to provide encrypted output frames; and a synchronizer (219) configured to substitute in a predetermined manner synchronization information corresponding to an encryption state of the crypto processor for a portion of the encrypted data in a portion of the encrypted output frames to provide resultant output synchronization frames suitable for synchronizing a decryption process at a target communication unit.

Abstract: A telephone (310) and a method for providing outbound audio when the telephone is operating in a speakerphone mode. A first data unit (350) including a first unit type identifier (360) can be received by the telephone. The first unit type identifier can be an indicator of a type of audio data contained in the first data unit. For instance, the first unit type identifier can indicate whether the audio data is music or non-music audio data. If the first unit type identifier has a first value, for example a value indicating that the audio data is music data, unmuted outbound audio reproduced from the first data unit can be provided and voice activity detection can be disabled. Additionally, inbound audio can be muted.

Abstract: The invention concerns a method (300) and system (100) for improving voice quality of a vocoder (138, 158). The method includes the steps of monitoring (312) a pitch of a voice signal (400) at a transmitting unit (110); when the pitch of the voice signal reaches a predetermined threshold (840), shifting (326) the pitch of the voice signal to at least a portion of a predetermined range (810); transmitting (338) the pitch-shifted voice signal to a receiving unit (112); and at the receiving unit, reshifting (342) the pitch-shifted voice signal to a level that compensates the step of shifting the pitch of the voice signal at the transmitting unit.

Abstract: A method of scaling polyphony can include identifying music data to be played (415), wherein the music data indicates instruments to be used and each instrument has an assigned priority. A measure of polyphony needed to play the music data can be compared with polyphony of a sound generating device (425). If the measure of polyphony exceeds the polyphony of the sound generating device, the music data can be played without using one or more instruments indicated by the music data according to the assigned priorities (440, 460).

Abstract: A speech filter (108) enhances the loudness of a speech signal by expanding the formant regions of the speech signal beyond a natural bandwidth of the formant regions. The energy level of the speech signal is maintained so that the filtered speech signal contains the same energy as the pre-filtered signal. By expanding the formant regions of the speech signal on a critical band scale corresponding to human hearing, the listener of the speech signal perceives it to be louder even though the signal contains the same energy.

Abstract: A system, wireless device (102) and method improve speaker intelligibility in a multi-party call by receiving a plurality of individual voice signals, determining a pitch contour for each individual voice signal, determining that the pitch contours for at least two of the individual voice signals are within a predetermined range relative to each other, and shifting the pitch of at least one voice signal a predetermined amount for the duration of the call. The pitch of the individual voice is shifted one to approximately five semitones. The method is performed at a central control station (110) prior to summation of the signals, or at an individual receiving unit (204) when three or more wireless devices (102) are communicating without the use of a central control station (110).

Abstract: Systems, an apparatus, and methods are provided for mitigating noise associated with an audio signal. A system (100) for mitigating noise associated with an audio signal includes an estimator module (108). The estimator module determines an estimated level of noise associated with the audio signal. The system also includes an expander module (110). The expander module causes an attenuation of the audio signal if a level of the audio signal is below a signal threshold. The expander module is adaptively tunable so that the attenuation caused (606) by the expander module is based upon the level of noise estimated (602) by the estimator module.

Abstract: Systems and methods for pre-conditioning a received audio signal prior to audio equalization of the audio signal are provided. The system (100) includes a spectral tilt estimator (110) for estimating a spectral tilt of the received audio signal, and a compensative filter synthesizer (115) for synthesizing a compensative filter based upon the spectral tilt estimated by the spectral tilt estimator. The filter comprises at least one compensative filter coefficient for mitigating the spectral tilt of the received audio signal prior to audio equalization of the audio signal.

Abstract: Systems (100 or 300) and methods (400 or 500) are provided for selecting a post-compression waveform from a post-compression waveform table (106) and supplying it to a synthesis engine (108). The post-compression waveform is based upon a set of post-compression coefficients determined by generating a frequency-domain representation of a periodic signal, the representation including at least one pre-compression frequency-domain sample (204), and performing a threshold-based compression of the pre-compression frequency-domain samples. Systems and methods also include indexing and storing (502) post-compression coefficients in a post-compression coefficient table (102), generating (506) a post-compression waveform based upon the set of post-compression coefficients, and placing (508) the post-compression waveform in the table prior to the selecting (510).

Abstract: A telephone (310) and a method for providing outbound audio when the telephone is operating in a speakerphone mode. A first data unit (350) including a first unit type identifier (360) can be received by the telephone. The first unit type identifier can be an indicator of a type of audio data contained in the first data unit. For instance, the first unit type identifier can indicate whether the audio data is music or non-music audio data. If the first unit type identifier has a first value, for example a value indicating that the audio data is music data, unmuted outbound audio reproduced from the first data unit can be provided and voice activity detection can be disabled. Additionally, inbound audio can be muted.

Abstract: A method (400, 600) and apparatus (500, 800) allow adjusting parameters associated with an audio signal output from a device. Tones are output according to a first interactive test profile presented on a user interface during a first test period. A first interaction forms a first adjustment profile with adjustment levels. The audio signal is output with parameters adjusted according to the first adjustment profile. The audio signal is output from the device in accordance with the first adjustment profile and a second interactive test profile. An interaction with the user interface forms a second adjustment profile having second adjustment levels and the audio signal is then adjusted in accordance with the second adjustment levels. The second interactive test profile includes a speech sample and an intelligibility parameter including a spectral tilt for the device and a formant sharpening profile.

Abstract: An apparatus (213) and corresponding methods (FIG. 7) to facilitate maintaining crypto synchronization while processing communication signals in a communication unit includes a vocoder (215) configured to convert input audio band signals to vocoder output frames; a crypto processor (217) configured to encrypt the vocoder output frames to provide encrypted output frames; and a synchronizer (219) configured to substitute in a predetermined manner synchronization information corresponding to an encryption state of the crypto processor for a portion of the encrypted data in a portion of the encrypted output frames to provide resultant output synchronization frames suitable for synchronizing a decryption process at a target communication unit.

Abstract: The invention concerns a method (300) and system (100) for improving voice quality of a vocoder (138, 158). The method includes the steps of monitoring (312) a pitch of a voice signal (400) at a transmitting unit (110); when the pitch of the voice signal reaches a predetermined threshold (840), shifting (326) the pitch of the voice signal to at least a portion of a predetermined range (810); transmitting (338) the pitch-shifted voice signal to a receiving unit (112); and at the receiving unit, reshifting (342) the pitch-shifted voice signal to a level that compensates the step of shifting the pitch of the voice signal at the transmitting unit.

Abstract: A method, system and computer readable medium for increasing the audio perceptual loudness includes shifting at least one frequency of a first audio signal to create a second audio signal so as to increase the audio perceptual loudness. The power level of the second audio signal is not more than a power level of the first audio signal. The method also includes generating high-audio perceptual loudness tone alert sequences based on psychoacoustic and audiometric data. It further includes acquiring a listener's threshold audio profile; adding the listener's audio profile to the loudness sensitivity curve for producing the listener's tonal sensitivity curve; determining a required dB scaling for critical band tones from the listener's tonal sensitivity curve; normalizing the tonal sensitivity curve for creating a decibel curve; selecting a frequency range of the tones by using the tonal sensitivity curve; and spacing the sequence of tones along a critical band scale.

Abstract: A method of scaling polyphony can include identifying music data to be played (415), wherein the music data indicates instruments to be used and each instrument has an assigned priority. A measure of polyphony needed to play the music data can be compared with polyphony of a sound generating device (425). If the measure of polyphony exceeds the polyphony of the sound generating device, the music data can be played without using one or more instruments indicated by the music data according to the assigned priorities (440, 460).

Abstract: A method (200) of voice identification within a mobile communication device can include detecting a voice signal within the mobile communication device (225) and determining at least one voice feature of the voice signal (230). At least one of the voice features of the voice signal can be compared with voice profiles accessible by the mobile communication device (235). Each voice profile also can be associated with an identity. Accordingly, at least one of the voice features of the voice signal can be matched with one of the voice profiles (240) and the identity associated with the matched voice profile can be presented through the mobile communication device (255).

Abstract: A system, method and computer readable medium for adjusting volume levels of a Musical Instrument Digital Interface (MIDI) sound file for optimizing play on a sound device. The method on an information processing system includes calculating a first set of loudness levels for each instrument in a MIDI sound file and calculating a second set of loudness levels corresponding to an audio output range of a sound device. The method further includes generating a mapping between the first set of loudness levels and the second set of loudness levels corresponding to the audio output range of the sound device. The method further includes generating a gain term for each note in the MIDI sound file and modifying the MIDI sound file using the second set of loudness levels and the gain term for each note in the MIDI sound file.

Abstract: Human hearing perceives loudness based on critical bands corresponding to different frequency ranges. As a sound's frequency spectrum increases beyond a critical band into a pr3eviously unexcited critical band, the perception is that the sound has increased in loudness. To take advantage of this principle, a filter is applied to a speech signal so as to expand the formant bandwidths of formants in the speech sample.