An example may include detecting, via a controller, one or more microphones and one or more speakers in an area, measuring, via the one or more microphones, an initial frequency response of an audio signal generated by the one or more speakers inside the area and generating an initial room performance rating, comparing the initial frequency response to a target frequency response, creating audio compensation values to apply to the one or more speakers based on the comparison, and applying the audio compensation values to the one or more speakers.

An example may include detecting, via a controller, one or more microphones and one or more speakers in an area, measuring, via the one or more microphones, an initial frequency response of an audio signal generated by the one or more speakers inside the area and generating an initial room performance rating, comparing the initial frequency response to a target frequency response, creating audio compensation values to apply to the one or more speakers based on the comparison, and applying the audio compensation values to the one or more speakers.

An ordering system for automated processing of customer orders for a retail establishment. In some embodiments, an ordering device is disposed at an ordering location such as outside a building of the retail establishment. The ordering device is configured to generate a first audio stream responsive to an interaction with an on-site customer adjacent the ordering device. An on-site controller device includes an artificial intelligence engine configured to generate content responsive to the first audio stream. The artificial intelligence engine combines the generated content with a second audio stream from an on-site employee to transmit a seamless third audio stream, via the ordering device, to the on-site customer. The generated content may be further tailored based on one or more traits of the customer as detected by a sensor.

A speech intelligibility enhancing system for enhancing speech, the system comprising: a speech input for receiving speech to be enhanced; an enhanced speech output to output the enhanced speech; and a processor configured to convert speech received by the speech input to enhanced speech to be output by the enhanced speech output, the processor being configured to: extract a portion of the speech received by the speech input; calculate the power of the portion; estimate a contribution due to late reverberation to the power of the portion of the speech when reverbed; calculate a target late reverberation power; determine a time t.sub.i for the estimated contribution due to late reverberation to decay to the target late reverberation power; calculate a pause duration, wherein the pause duration is calculated using the time t.sub.i; insert a pause having the calculated duration into the speech received by the speech input at a first location, wherein the first location is followed by the portion.

An apparatus for processing a speech signal is provided. The apparatus includes a communicator comprising communication circuitry configured to transmit and receive data, an actuator comprising actuation circuitry configured to generate vibration and to output a signal, a formant enhancement filter configured to increase a formant of the speech signal, and a controller comprising processing circuitry configured to control the speech signal to be received through the communicator, to estimate at least one formant frequency from the speech signal based on linear predictive coding (LPC), to estimate a bandwidth of the at least one formant frequency, to determine whether the speech signal is a voiced sound or a voiceless sound, to configure the formant enhancement filter based on the at least one formant frequency, the bandwidth of the at least one formant frequency, characteristics of the determined voiced sound or voiceless sound, and signal delivery characteristics of a human body, to apply the formant enhancement filter to the speech signal, and to control the speech signal to which the formant enhancement filter is applied to be output using the actuator through the human body.

An apparatus for processing a speech signal is provided. The apparatus includes a communicator comprising communication circuitry configured to transmit and receive data, an actuator comprising actuation circuitry configured to generate vibration and to output a signal, a formant enhancement filter configured to increase a formant of the speech signal, and a controller comprising processing circuitry configured to control the speech signal to be received through the communicator, to estimate at least one formant frequency from the speech signal based on linear predictive coding (LPC), to estimate a bandwidth of the at least one formant frequency, to determine whether the speech signal is a voiced sound or a voiceless sound, to configure the formant enhancement filter based on the at least one formant frequency, the bandwidth of the at least one formant frequency, characteristics of the determined voiced sound or voiceless sound, and signal delivery characteristics of a human body, to apply the formant enhancement filter to the speech signal, and to control the speech signal to which the formant enhancement filter is applied to be output using the actuator through the human body.

In a communication apparatus, an encoder compresses telephone call voice which is transmitted from another communication apparatus. A voice accumulator preserves the telephone call voice, which is compressed by the encoder, as a message. A decoder expands the telephone call voice which is preserved in the voice accumulator. A signal memory temporarily maintains the telephone call voice which is expanded by the decoder. A speech speed convertor performs speech speed conversion on the telephone call voice, which is read from the signal memory, and outputs resulting voice from a speaker. A memory monitor temporarily stops to expand the telephone call voice in the decoder in a case where the memory monitor determines that an idle capacity of the signal memory approaches a predetermined lower limit value.

In a communication apparatus, an encoder compresses telephone call voice which is transmitted from another communication apparatus. A voice accumulator preserves the telephone call voice, which is compressed by the encoder, as a message. A decoder expands the telephone call voice which is preserved in the voice accumulator. A signal memory temporarily maintains the telephone call voice which is expanded by the decoder. A speech speed convertor performs speech speed conversion on the telephone call voice, which is read from the signal memory, and outputs resulting voice from a speaker. A memory monitor temporarily stops to expand the telephone call voice in the decoder in a case where the memory monitor determines that an idle capacity of the signal memory approaches a predetermined lower limit value.

Embodiments herein include receiving a request to modify an audio characteristic associated with a first user for a voice communication system. One or more suggested modified audio characteristics may be provided for the first user, based on, at least in part, one or more audio preferences established by another user. An input of one or more modified audio characteristics may be received for the first user for the voice communication system. A user-specific audio preference may be associated with the first user for voice communications on the voice communication system, the user-specific audio preference including the one or more modified audio characteristics.

The present invention is a computer program product and method for increasing the playback speed of audio or other media files. The computer program product and method identifies pedagogic media files and adds a flag to the metadata of the media file. The flag represents the number and type of pauses or silent sections in the pedagogic media file. Based on the flag, the computer program product and method may fast forward or remove a portion of the pauses and silent sections to provide a new playback speed.