18results about "Automatic tone/bandwidth control" patented technology

A computer system for manipulating, combining, or composing dynamic sounds accesses a package of one or more music stems. The computer system then receive an input variable from one or more vehicle sensors. The one or more vehicle sensors measure an aspect of driving parameters of a vehicle. In response to the input variable, the computer system generate a particular audio effect with the one or more music stems.

In various embodiments, the disclosed techniques automatically change the audio equalization levels of audio content. When an audio content is requested for playback, the equalization manager of an audio system extracts the frequency components of a segment of the audio content. The equalization manager analyzes the frequency components of the segment and separates the frequency components into predefined frequency bands based on the analysis. The equalization manager determines sound pressure level (SPL) metrics of each frequency component. The equalization manager generates a custom equalization setting using gains calculated for each frequency band based on the SPL metrics. The equalization manager then adjusts the audio segment by applying the custom equalization setting to generate a modified segment (e.g., applying the calculated gains to each of the frequency bands). The modified segment is then output as audio to a user. The process then repeats for additional segments of the audio.

In various embodiments, the disclosed technology automatically alters audio equalization levels of audio content. When a request to play audio content is made, an equalization manager of an audio system extracts frequency components of a segment of the audio content. The equalization manager analyzes the frequency components of the segment and separates the frequency components into predefined frequency bands based on the analysis. The equalization manager determines a sound pressure level (SPL) indicator for each frequency component. The equalization manager generates a customized equalization setting based on the SPL indicators using gains computed for each frequency band. The equalization manager then adjusts the audio segment by applying the customized equalization setting to generate a modified segment (e.g., applying the computed gains to each of the frequency bands). The modified audio segment is then output as audio to a user. The process is then repeated for additional segments of the audio.

A method of equalising a signal derived from an input device, the method comprising: receiving the signal; applying an order-statistic filter to the signal in the frequency domain to generate a statistically filtered signal; equalising the received signal based on the statistically filtered signal to generate an equalised signal.

A signal processing system for adjusting an audio signal, the signal processing system configured to: receive an audio signal for adjustment at a signal input; at a function generator, receive a selection of a musical key and scale, the musical key and scale including musical notes, and generate an audio signal processing function, configured to adjust the amplitude of frequencies of an audio signal, the audio signal processing function being dependent on the selected musical key and scale; apply the generated audio signal processing function to the received audio signal to generate an adjusted audio signal; and, output the adjusted audio signal at a signal output.

A system is described that comprises a computing device comprising one or more processors and a memory. The computing device is configured to receive user input and to analyze the user input to generate control data. The computing device is further configured to analyze the audio input to generate signal data that are characteristic of the audio input and to analyze previous tuning data to generate reference data. The computing device is also configured to process at least the control data, the signal data, and reference data to generate tuning data. In addition, the computing device is configured to process audio input to generate audio output based on at least one audio model and the tuning data.

A system is described that comprises a computing device having one or more processors and a memory. The computing device is configured to receive a first stream of first data. The computing device is further configured to perform an analysis on the first stream of first data at a first time and at a second time that is different from the first time and determine whether there is a change in the first stream of first data between the first time and the second time. In addition, the computing device is configured to, if there is a change in the first stream of first data between the first time and the second time, adjust at least one tuning parameter of an audio tuning algorithm of an audio processing system. The audio processing system is configured to process audio input by means of the audio tuning algorithm to generate audio output.

According to the present invention there is provided an artificial voice generation system designed to restore or augment speech for individuals experiencing voice loss or desiring an alternative vocal quality. The system comprises a controllable air pump, an airflow member delivering air to the user's oral cavity, and a sound generation member— such as a replaceable membrane cartridge—configured to vibrate and produce sound in response to airflow. A user interface allows manual or automatic modulation of airflow and voice parameters, enabling real-time adjustment of pitch, loudness, and voice quality, including male, female, or non-binary characteristics. The system may include sensors for pressure and airflow, anti-jamming features, and can be adapted for use with or without a neck stoma, making it suitable for a wide range of users, from laryngectomy patients to those with temporary or chronic voice loss. By providing a customisable, high-quality artificial voice source with improved naturalness and intelligibility, the invention addresses limitations of existing devices such as electrolarynxes and tracheoesophageal prostheses, offering a versatile and user-friendly solution for voice rehabilitation and enhancement.

The application discloses a high-fidelity AB class audio power amplifier. According to an embodiment of the audio power amplifier, the main architecture is a two-stage operational amplifier adopting a gain boosting technology, and according to a circuit topology structure and working principle, the audio power amplifier can be divided into five parts, i.e. an input stage, a gain boosting stage, an output stage, a bias circuit of a floating voltage source and a frequency compensation circuit. By setting the gain boosting stage, the output impedance can be increased without additionally increasing the common-source and common-gate device, and then the gain is increased. The application can make the audio power amplifier keep high gain to reduce nonlinearity when driving a small resistance earphone load, and a complex frequency compensation network is not needed to be designed.

A system is described that includes a computing device having one or more processors and memory. The computing device is configured to receive a first stream of first data. The computing device is further configured to perform an analysis of the first stream of first data at a first time and a second time different from the first time, and determine whether the first stream of first data changes between the first time and the second time. Additionally, the computing device is configured to adjust at least one adjustment parameter of an audio adjustment algorithm of an audio processing system if the first stream of first data changes between the first time and the second time. The audio processing system is configured to process an audio input through the audio adjustment algorithm to generate an audio output.

A system comprising audio processing circuitry is provided. The audio processing circuitry is operable to receive audio signals. The audio processing circuitry is operable to process the audio signals to detect strength of a chat component of the audio signals and strength of a game component of the audio signals. The audio processing circuitry is operable to automatically control a volume setting based on one or both of: the detected strength of the chat component, and the detected strength of the game component. The combined-game-and-chat audio signals may comprise a left channel signal and a right channel signal. The processing of the combined-game-and-chat audio signals may comprise measuring strength of a vocal-band signal component that is common to the left channel signal and the right channel signal.

Methods, apparatus, systems and articles of manufacture are disclosed for playback using pre-processed profile information and personalization. Example apparatus disclosed herein include a synchronizer to, in response to receiving a media signal to be played on a playback device, access an equalization (EQ) profile corresponding to the media signal; an EQ personalization manager to generate a personalized EQ setting; and an EQ adjustment implementor to modify playback of the media signal on the playback device based on a blended equalization generated based on the EQ profile and the personalized EQ setting.

A method comprises outputting at least one test signal (Sn) to a listening environment (30) by means of a loudspeaker (10), receiving each of the at least one test signal (Sn) by means of a microphone (22) arranged in the listening environment (30), determining at least one of a position and an orientation of the microphone (22) in the listening environment (30) with respect to the loudspeaker (10) at the time of receipt of each of the at least one test signal (Sn), evaluating the test signals (Sn) received by the microphone (22), and determining at least one aspect related to at least one transfer function based on the evaluation of the test signal (Sn) and on at least one of the position and the orientation of the microphone.

A network transceiver device is provided, including at least two variable gain amplifiers (VGAs), and at least two sets of analog digital converters (ADCs), each set including ADCs coupled to an output of one of the VGAs, the sets being arranged in VGA-specific channels. The device includes a plurality of feed-forward equalizers (FFEs), each FFE being coupled to receive an output of one of the ADCs in one of the VGA-specific channels. Each FFE is configured to adaptively equalize the output received from the ADCs utilizing a first equalization coefficient subset with coefficient values that are common to all FFEs, and a second equalization coefficient subset that is channel specific and that has a first set of coefficient values for a first VGA-specific channel and a second set of coefficient values for a second VGA-specific channel, the sets of coefficient values being computed independently.

The present disclosure relates to methods and apparatuses to set the bulk bias of resistors. An example apparatus includes a first resistor (130, 245, 305, 405, 505) having a first terminal and a bias terminal, a second resistor (140, 250, 410, 510) having a first terminal and a bias terminal, and bias circuitry (150, 225, 310, 415, 515) including a third resistor (520, 710) having a first terminal and a second terminal, the first terminal of the third resistor coupled to the first terminal of the first resistor, a fourth resistor (525, 720) having a first terminal and a second terminal, the first terminal of the fourth resistor coupled to the bias terminal of the first resistor and the second terminal of the third resistor, a fifth resistor (530, 730) having a first terminal and a second terminal, the first terminal of the fifth resistor coupled to the first terminal of the second resistor, a sixth resistor (535, 740) having a first terminal coupled to the bias terminal of the second resistor and the second terminal of the fifth resistor, and buffer circuitry (540).