Abstract: Systems and methods for automatically adjusting volume in a vehicle are provided. One method includes generating a noise prediction based on images received from a vehicle camera; determining a cabin noise level based on the noise prediction and an audio signal detected by a cabin microphone; and selectively amplifying a quiet passage within the detected audio signal to above the cabin noise level while maintaining volume level of remaining passages within the detected signal. A vehicle system includes a cabin microphone for detecting an audio signal; a camera for capturing images; and a processor configured to: generate a noise prediction based on the images; determine a cabin noise level based on the noise prediction and the detected audio signal; and selectively amplify a quiet passage within the detected signal to above the cabin noise level while maintaining volume level of remaining passages within the detected signal.

Abstract: An electronic device (200) includes a user interface (205), a communication circuit (206), and an audio capture device (212) to receive ambient audio signals from an environment about the electronic device. A voice control interface engine (214) is operable with the audio capture device to process signals received from the audio capture device to identify at least some speech from the ambient audio signals. One or more processors (203) are operable with the voice control interface engine to compute a noise index as a function of the ambient audio signals and the at least some speech. The one or more processors can optionally identify a source of the at least some speech. Where the noise index exceeds a predefined criterion, the one or more processors can execute a communication operation on one or more of the user interface or the communication circuit.

Abstract: Disclosed is a headset including at least one in-ear microphone; at least one out-ear microphone; a control unit including an in-ear signal processing module for extracting low frequency components from a signal sensed through the in-ear microphone, an out-ear signal processing module for extracting high frequency components from a signal sensed through the out-ear microphone, and a mixing module for mixing the extracted low frequency components and high frequency components and outputting the mixed signal; and a communication unit for transmitting the signal output from the mixing module of the control unit to an external device, and sound quality can be improved by removing noises in a voice signal of a user transferred through the headset and creating a signal close to the voice of the user.

Abstract: A interface circuit for an acoustic transducer provided with a first detection structure and a second detection structure has: a first input and a second input; a first processing path and a second processing path coupled, respectively, to the first input and second input and supply a first processed signal and a second processed signal; and a recombination stage, which supplies a mixed signal by combining the first processed signal and the second processed signal with a respective weight that is a function of a first level value of the first processed signal. The first and second inputs receive a respective detection signal associated, respectively, to the first detection structure and to the second detection structure of the acoustic transducer; and an output stage the first processed signal, the second processed signal or the mixed signal, on the basis of a second level value of the first processed signal.

Abstract: A method for providing elected sounds to a user—isolated from sounds of the environment—of a mobile device to increase his awareness regarding entities or events in his vicinity. The user wears sound isolating headphones connected to his mobile device. A mobile application is installed on the mobile device, and is automatically activated when the headphones are connected to his mobile device. The application is adapted to automatically activate the device's microphone when the isolating headphones are plugged into the device and periodically compare in real-time, sounds received from the environment, to a predefined collection of reference sounds. Sounds of the environment, which match one or more reference sounds from the collection are selectively filtered out and as long as the sounds received from the environment match the one or more reference sounds from the collection, the filtered sounds are continuously passed to the isolating headphones.

Abstract: Audio systems for a vehicle and methods for increasing auditory situation awareness in a vehicle are provided. An audio system includes at least one ambient microphone disposed on the vehicle, a processor and at least one loudspeaker. The at least one ambient microphone is configured to capture ambient sound external to the vehicle and to produce an ambient sound signal. The processor is configured to receive the ambient sound signal and an audio content signal, and to mix the ambient sound signal with the audio content signal to generate a mixed output signal. The at least one loudspeaker is configured to reproduce the mixed output signal in the vehicle cabin.

Abstract: An apparatus and method are disclosed, which separates ambient noise from a voice signal in a Bluetooth headset with dual microphones, switches the connection path between the dual microphones and automatically controls output audio gain, based on the ambient noise. The method for outputting the voice signal includes receiving/comparing the first and second input signals from the first and second microphones, extracting the voice signal of substantially identical intensity from the first and second input signals, and transmitting the voice signal from the portable terminal communicating with the Bluetooth headset.

Abstract: Disclosed herein are methods, systems, and devices for mitigating the impact of noise, such a low-frequency noise caused by wind, on sounds received at a hearing prosthesis. An example method includes receiving an external sound signal transduced externally to a recipient from an ambient sound and an internal sound signal transduced internally to the recipient from the ambient sound. The example method also includes determining that a triggering condition is present in the external sound signal. The triggering condition may be indicative of a condition in that more adversely affects externally-received sounds than internally-received sounds. In response to determining that the triggering condition is present in the external sound signal, the example method further includes generating a stimulation signal that is based at least in part on spectral information of the internally-transduced sound.

Abstract: The present invention concerns a MEMS Microphone (1) comprising, a transducer element (2) for providing an electrical signal, a first part (3) for receiving the electrical signal from the transducer element (2) and for providing a processed signal, a second part (4) for receiving the processed signal from the first part (3) and for providing an output signal of the MEMS microphone (1), and a gain control unit (5) that is enabled to adjust a gain setting of the first part (3) and to adjust a gain setting of the second part (4). Further, another aspect of the present invention concerns a method of operating said MEMS microphone (1) comprising the step of adjusting a gain setting of the first part (3) and adjusting a gain setting of the second part (4).

Abstract: A mobile communication terminal is disclosed. The mobile communication terminal has Receiver Mode and hands-free mode and switchable between the modes. The terminal includes a first sound generator, which works under Receiver Mode and used for receiving and playing voice signal; a second sound generator, which works under Receiver Mode and Hands-free Mode and used for receiving and playing voice signal; when the mobile communication terminal works under Receiver Mode, the first sound generator and the second sound generator receive and play voice signal at the same time. A method of improving sound quality of the mobile communication terminal is also provided.

Abstract: Multipath digital microphone systems comprising a multipath digital audio combiner component are described. Exemplary multipath digital microphone systems can switch from conveying one digital audio signal to conveying another digital audio signal based on one or more switching criteria determined by an exemplary multipath digital audio combiner component. Provided implementations can be configured to switch from conveying one digital audio signal to conveying digital audio signal according to an algorithm to provide low-power, high dynamic range digital microphone systems, without audible artifacts associated with conventional digital microphone systems.

Abstract: A wireless communication device and method for same, including: a first communication unit configured to use a non-IP-based wireless communication protocol, a second communication unit configured to use an IP-based wireless communication protocol, and a controller configured to: search for a wireless device for communication, the wireless communication device configured to provide a universal plug and play (UPnP) service, the search being by use of the first communication unit, and initiate the UPnP service with respect to a found wireless device for communication by use of the second communication unit.

Abstract: A method for controlling a gain applied to an audio signal. The method comprises applying a gain step to a gain setting at defined time intervals, and upon detection of at least one zero crossing point within the audio signal, applying the gain setting to a gain control signal for controlling the gain applied to the audio signal.

Abstract: According to an embodiment, a method of sensing motion includes receiving a first signal from a first pressure sensor and a second signal from a second pressure sensor, comparing the first signal and the second signal, and characterizing a motion based on the comparing.

Abstract: An interface for expanding a signal starting from a first sensing signal and a second sensing signal, wherein a receiving intensity measuring element generates an intensity signal; and a selector is controlled to select each time the first sensing signal, the second sensing signal, or a combined signal deriving from a weighted combination of these signals. The selector uses a plurality of thresholds variable as a function of the intensity signal.

Abstract: Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.

Abstract: An audio mixer for mixing a plurality of audio tracks to a mixture signal has a semantic command interpreter for receiving a semantic mixing command and for deriving a plurality of mixing parameters for the plurality of audio tracks from the semantic mixing command; an audio track processor for processing the plurality of audio tracks in accordance with the plurality of mixing parameters; and an audio track combiner for combining the plurality of audio tracks processed by the audio track processor into the mixture signal. A corresponding method has: receiving a semantic mixing command; deriving a plurality of mixing parameters for the plurality of audio tracks from the semantic mixing command; processing the plurality of audio tracks in accordance with the plurality of mixing parameters; and combining the plurality of audio tracks resulting from the processing of the plurality of audio tracks to form the mixture signal.

Abstract: Systems and methods may provide for determining a usage context of a headset and detecting one or more danger-related conditions based on the usage context. Additionally, one or more settings of the headset may be adjusted in response to at least one of the one or more danger-related conditions. In one example, adjusting the one or more settings includes one or more of deactivating noise cancellation, adjusting noise cancellation to increase an intensity of at least one of the one or more danger-related sounds, or adjusting one or more audio playback settings associated with the headset.

Abstract: A system and method for dynamic residual noise shaping configured to reduce hiss noise in an audio signal. The system and method may detect an amount and type of hiss noise. The system and method may limit calculated noise suppression gains responsive to the detected amount and type of hiss noise. The limited noise suppression gains may be applied to the audio signal and may reduce the hiss noise.

Abstract: Embodiments are described for an in-vehicle computing system, and methods of controlling vehicle settings using the in-vehicle computing system based on input received from a wearable device. Input may also be received from a mobile device. The in-vehicle computing system may automatically adjust one or more vehicle settings based on the received input.

Abstract: Systems and methods for dynamically adjusting transmit gain in a transceiver. The gain is adjusted in order to provide the maximum gain. The amount of distortion is measured. The gain is increased until the distortion reaches a predetermined limit. The gain of several components can be adjusted independently.

Abstract: A headset and a method configured to process audio signals from multiple microphones, comprising: a first pair of microphones (101,102) outputting a first pair of microphone signals and a second pair of microphones (103, 104) outputting a second pair of microphone signals; a first near-field beamformer (105) and a second near-field beamformer (106) each configured to receive a pair of microphone signals and adapt the spatial sensitivity of a respective pair of microphones as measured in a respective beamformed signal (XL; XR) output from a respective beamformer (105; 106); wherein the spatial sensitivity is adapted to suppress noise relative to a desired signal; a third beamformer (107) configured to dynamically combine the signals (XL; XR) output from the first beamformer (105) and the second beamformer (106) into a combined signal (XC); wherein the signals are combined such that signal energy in the combined signal is minimized while a desired signal is preserved; and a noise reduction unit (109) configured

Abstract: The invention discloses a headset communication method under a strong-noise environment and a headset.

Abstract: A interface circuit for an acoustic transducer provided with a first detection structure and a second detection structure has: a first input and a second input; a first processing path and a second processing path coupled, respectively, to the first input and second input and supply a first processed signal and a second processed signal; and a recombination stage, which supplies a mixed signal by combining the first processed signal and the second processed signal with a respective weight that is a function of a first level value of the first processed signal. The first and second inputs receive a respective detection signal associated, respectively, to the first detection structure and to the second detection structure of the acoustic transducer; and an output stage the first processed signal, the second processed signal or the mixed signal, on the basis of a second level value of the first processed signal.

Abstract: Thus there is provided an earphone or headset comprising a housing, an internal microphone, an external microphone and a signal processing unit. The signal processing unit is connected to the internal microphone and the external microphone. The signal processing unit serves for signal processing of an audio signal of the external microphone and for adding the signal-processed audio signal to an audio signal from the internal microphone. The signal processing unit is adapted to perform band pass limitation of the audio signal of the external microphone and to output the added signal as an output signal.

Abstract: In one aspect, the present invention facilitates the investigation of networks of criminals, by gathering associations between phone numbers, the names of persons reached at those phone numbers, and voice print data. In another aspect the invention automatically detects phone calls from a prison where the voiceprint of the person called matches the voiceprint of a past inmate. In another aspect the invention detects identity scams in prisons, by monitoring for known voice characteristics of likely imposters on phone calls made by prisoners. In another aspect, the invention automatically does speech-to-text conversion of phone numbers spoken within a predetermined time of detecting data indicative of a three-way call event while monitoring a phone call from a prison inmate. In another aspect, the invention automatically thwarts attempts of prison inmates to use re-dialing services.

Abstract: An adaptive noise canceling (ANC) circuit adaptively generates an anti-noise signal from a reference microphone signal that is injected into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone proximate the speaker provides an error signal. A secondary path estimating adaptive filter estimates the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. Tones in the source audio, such as remote ringtones, present in downlink audio during initiation of a telephone call, are detected by a tone detector using accumulated tone persistence and non-silence hangover counting, and adaptation of the secondary path estimating adaptive filter is halted to prevent adapting to the tones. Adaptation of the adaptive filters is then sequenced so any disruption of the secondary path adaptive filter response is removed before allowing the anti-noise generating filter to adapt.

Abstract: An acoustic device, which reproduces an acoustic signal of one acoustic source selected from a plurality of acoustic sources. A first averaging unit averages signal levels of the acoustic signal for a first time period to obtain a first average signal. A second averaging unit averages the signal levels for a second time period shorter than the first time period to obtain a second average signal. A selecting unit selects either the first average signal or the second average signal as one signal having a signal level higher than the other. A gain decreasing unit decreases a gain of the acoustic signal according to the signal level of the one signal if the signal level of the one signal is higher than a threshold value determined according to the one acoustic source. A correcting unit corrects the signal levels of the acoustic signal according to the gain.

Abstract: An interior rearview mirror system suitable for use in a vehicle includes an interior rearview mirror assembly having a mirror head and a reflective element. The mirror head includes a first microphone operable to generate a first analog signal and a second microphone operable to generate a second analog signal. The first analog signal is converted to a first digital signal by at least one analog to digital converter and the second analog signal is converted to a second digital signal by the at least one analog to digital converter. A digital sound processor is operable to process the first and second digital signals. Responsive to the processing of the first and second digital signals, the digital sound processor generates a digital output, and the digital output, at least in part, distinguishes a human voice present in the vehicle from noise present in the vehicle.

Abstract: Provided is a method for adaptively enhancing an end-user's perceived quality, or quality of experience (QoE), of speech and other audio under ambient noise conditions. The method comprises the steps of determining the ambient noise characteristics on a continuous basis to capture the time varying nature of ambient noises, and adaptively determining the most optimal signal shaping to be applied to the audio/speech signal to produce the most appropriate enhancement to compensate for the ambient noise impairment. The method also comprises a signal shaping technique by using an infinite impulse response (IIR) filter that performs the signal modification with a low delay; a multi-level automatic gain control (AGC); and a controlled amplitude clipping module that assures samples are below a certain limit; and outputs the modified signal for playback through a loudspeaker or the like.

Abstract: Provided is a computer program product for adaptively enhancing an end-user's perceived quality, or quality of experience (QoE), of speech and other audio under ambient noise conditions. The computer program product comprises the steps of determining the ambient noise characteristics on a continuous basis to capture the time varying nature of ambient noises, and adaptively determining the most optimal signal shaping to be applied to the audio/speech signal to produce the most appropriate enhancement to compensate for the ambient noise impairment. The computer program product also comprises a signal shaping technique by using an infinite impulse response (IIR) filter that performs the signal modification with a low delay; a multi-level automatic gain control (AGC); and a controlled amplitude clipping module that assures samples are below a certain limit; and outputs the modified signal for playback through a loudspeaker or the like.

Abstract: Active noise cancellation is combined with spectrum modification of a reproduced audio signal to enhance intelligibility.

Abstract: The present invention discloses a method and an implementation apparatus for intelligently controlling volume of an electronic device, where the electronic device is installed with microphones. The method includes receiving an external sound picked up by the microphones, extracting content information of the external sound, determining, according to the content information of the external sound, whether a prestored sound template database includes a sound that is matched with the content information of the external sound, and generating a first determining result. If the first determining result is yes, the method includes adjusting volume of a sound that is being output, or generating a prompt tone, turning off a sound that is being output, and outputting the prompt tone. In the present invention, an electronic device is enabled to intelligently control its own volume according to an external sound.

Abstract: A loudness enhancement system and method is described that increases the loudness of an audio signal being played back by an audio device that places limits on the dynamic range of the audio signal. In an embodiment, the loudness enhancement system and method compresses the audio signal to an adaptively-determined compression limit that is greater than or equal to a maximum desired output level and then applies an adaptively-determined degree of soft clipping to the compressed audio signal. The compression limit and degree of soft clipping may be determined based on an overload measure that is calculated for successive portions of the audio signal. The loudness enhancement system and method advantageously operates in a manner that generates less distortion than the method of simply over-driving the audio signal such that hard-clipping occurs.

Abstract: Systems and methods for audio difference detection include a memory component that stores logic that, when executed by a processor, causes a vehicle amplifier to receive a test input signal for a generic test sequence to determine a standard input signal for producing a desired output response signal, store the standard input signal with standard amplifier settings associated with the standard input signal, and receive an audio signal from a head unit of the vehicle, where the head unit is part of the vehicle audio system for the vehicle in which the vehicle amplifier is utilized. In some embodiments, the logic causes the vehicle amplifier to determine whether there is a difference between the audio signal and the standard input signal and in response to determining there is a difference, and adjust vehicle amplifier settings of the vehicle amplifier, such that vehicle speakers output the desired output response signal.

Abstract: Systems, devices and methods provide feedback about the quality of communication between a device and a remote control. A wireless signal is sent between a controlled device and the remote control. The quality of the signal can be measured and reported to a user, installer, troubleshooter, customer service agent or other person in any manner. Quality may be determined based upon the strength of the received signal as well as the amount of noise that is present. The quality measurements provide feedback that allows a user, installer, customer service representative or other person to change the positions of the device or the remote control, or to take other actions based upon the quality of the wireless signal that is received.

Abstract: The embodiments herein provide an apparatus and method for cancelling signal noise. According to one embodiment, an apparatus for cancelling signal noise has a sensor or receiver to capture the undesirable signals. A transducer converts the energy of the captured signals and modulates the captured undesirable signals. A signal inverting circuit is connected to the transducer to generate the inverse of the captured undesirable signals by inverting the amplitude of the undesirable signal while maintain the frequency at the same level. The generated inverse of the undesirable signal transmitted by a transmitter is received by a receiver and output through a speaker so that the output inverse of the undesirable signal is combined with the undesirable signal to produce a desired signal environment.

Abstract: An electronic device includes an audio sensor, an audio output unit, a pulse width modulation (PWM) unit connected to the audio output unit, and a control chip. When the audio output unit outputs the audio signal, the audio sensor detects a volume level of the environment sound. The control chip determines a volume level of the audio signal according to the volume level of the environment sound and preset associations between volume levels of the audio signal outputted by the audio output unit and volume levels of the environment, and determines a duty cycle of the PWM unit according to the determined volume level and the maximum volume level. The control chip then outputs the duty cycle to the PWM unit, to adjust a current volume level of the audio signal outputted by the audio output unit to the determined volume level.

Abstract: A system and method for dynamic residual noise shaping configured to reduce hiss noise in an audio signal. The system and method may detect an amount and type of hiss noise. The system and method may limit calculated noise suppression gains responsive to the detected amount and type of hiss noise. The limited noise suppression gains may be applied to the audio signal and may reduce the hiss noise.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: A method and apparatus for adaptively controlling the audio output (122) of a communication device (114) according to the noise characteristics of the receiver listening environment (119). An output volume for the communication device (114) is set by a user (e.g., listener 118). The communication device (114) can intermittently sample ambient noise (116) of its environment (119). A minimum signal to noise threshold can be established for audio output (122). A total adjustment for the audio output (122) is established based on the ambient noise (116), the user set output volume, and the minimum signal to noise threshold. The total adjustment is a product of a software volume adjustment (230) and a hardware gain adjustment (240). The software volume adjustment (230) and the hardware gain adjustment (240) is adaptively applied when the communication device (114) outputs audio (122).

Abstract: A close talk detection method and system is provided for active noise cancellation system on a cellular telephone or the like, based on two properly located microphones. Given the location of the microphones and the way people use the phone, the power ratio (difference) at the two microphones implies the location of the speaker within a given range. The improved close-talk detector is not affected by power levels or SNR of non-close-talk ambient disturbances. Power levels of both a voice and a reference microphone are measured and the ratio r of these power levels is determined. If the ratio r is greater than a predetermined threshold (e.g., 7 dB), then close talk is occurring. If the ratio r is less than the predetermined threshold, then the signal is determined to be loud ambient noise or some other non-close-talking signal and noise cancellation processing is not affected.

Abstract: An audio apparatus includes a sound generation unit configured to generate a measuring signal, a sound collection unit configured to obtain a sound collection signal, a detection unit configured to detect a level difference between the sound collection signal and background noise, a characteristic calculation unit configured to extract each signal period of the measuring signal from the sound collection signal, a feature amount calculation unit configured to calculate a feature amount representative of sudden noise for each period based on each of the period signals, and a determination unit configured to compare feature amounts of the periods and to remove, from periods to be used for averaging the period signals, a period or periods whose feature amounts are not within a range between a threshold value and a minimum value of the feature amounts as a reference, wherein the threshold value is determined according to the detected level difference.

Abstract: An apparatus and method for adjusting the volume in a portable terminal. The apparatus includes an information collection unit and a volume adjustment unit. The information collection unit collects noise judgment information that is used for judging the extent of background noise. The volume adjustment unit estimates background noise using the collected information, and adjusts the volume according to the background noise.

Abstract: The disclosed invention provides the ability for an operator, such as a sonar watch stander, to pay attention to both ambient sounds and system audio sound simultaneously. Further, the disclosed invention allows an operator to hear information being disseminated in the immediate environment, while still maintaining full sonar audio capability. The present invention allows the operator to take advantage of the benefits of binaural processing.

Abstract: System, method, and computer program product to provide targeted and variable information during a remote conference, by: responsive to a first person joining the remote conference, playing a first tone having a first acoustical characteristic indicating an arrival of the first person, responsive to a second person joining the remote conference, playing a second tone having a second acoustical characteristic indicating an arrival of the second person, the second acoustical characteristic different than the first, and responsive to one of the first person and the second person leaving the remote conference, playing a third tone having a third acoustical characteristic indicating a departure of one of the first person and the second person, the third acoustical characteristic different than the second, a type of the first, second, and third acoustical characteristics being the same, the first, second, and third tone audible to at least one person on the remote conference.

Abstract: System, method, and computer program product to provide targeted and variable information during a remote conference, by: responsive to a first person joining the remote conference, playing a first tone having a first acoustical characteristic indicating an arrival of the first person, responsive to a second person joining the remote conference, playing a second tone having a second acoustical characteristic indicating an arrival of the second person, the second acoustical characteristic different than the first, and responsive to one of the first person and the second person leaving the remote conference, playing a third tone having a third acoustical characteristic indicating a departure of one of the first person and the second person, the third acoustical characteristic different than the second, a type of the first, second, and third acoustical characteristics being the same, the first, second, and third tone audible to at least one person on the remote conference.

Abstract: The dynamic setting of an increment on an amplitude scale of an output signal includes receiving a user input requesting a change in an amplitude of the output signal. The amplitude of an external signal related to the output signal is determined, and a set of increments on the amplitude scale is calculated based upon the current amplitude of the output signal and the amplitude of the external signal. The existing set of increments are replaced with the newly calculated set of increments and the amplitude of the output signal is set to the next increment of the newly calculated set of increments.

Abstract: Controlling a multimedia software application using high-level metadata features and symbolic object labels derived from an audio source, wherein a first-pass of low-level signal analysis is performed, followed by a stage of statistical and perceptual processing, followed by a symbolic machine-learning or data-mining processing component is disclosed. This multi-stage analysis system delivers high-level metadata features, sound object identifiers, stream labels or other symbolic metadata to the application scripts or programs, which use the data to configure processing chains, or map it to other media. Embodiments of the invention can be incorporated into multimedia content players, musical instruments, recording studio equipment, installed and live sound equipment, broadcast equipment, metadata-generation applications, software-as-a-service applications, search engines, and mobile devices.

Abstract: A device and method are disclosed for testing the intelligibility of audio announcement systems. The device may include a microphone, a translation engine, a processor, a memory associated with the processor, and a display. The microphone of the analyzer may be coupled to the translation engine, which in-turn may be coupled to the processor, which is in-turn may be coupled to the memory and the display. The translation engine can convert audio speech input from the microphone into data output. The processor can receive the data output and can apply a scoring algorithm thereto. The algorithm can compare the received data against data that is stored in the memory of the analyzer and calculates the accuracy of the received data. The algorithm may translate the calculated accuracy into a standardized STI intelligibility score that is then presented on the display of the analyzer.