Abstract: An audio processing system and a method thereof are provided. The system includes an audio receiver and an audio processor device. The audio receiver includes an audio interface, an absolute value converter unit, a framing unit and a characteristic value detector unit. The audio interface receives an audio signal from an audio transceiver device. The absolute value converter unit takes an absolute value of the audio signal to output an absolute value audio signal. The framing unit divides the absolute value audio signal into a plurality of sub-frame signals. The characteristic value detector unit detects characteristic values of the sub-frame signals. The audio processer processes the audio signal outputted by the audio transceiver device according to the characteristic values of the sub-frame signals.

Abstract: An audio adjusting method and an audio adjusting device are provided. The audio adjusting method includes the steps of: receiving a plurality of directional audio signals; calculating a displacement amount through a positioning module; calculating a plurality of gain values corresponding to the displacement amount through a processing module; adjusting volume levels of the plurality of directional audio signals according to the plurality of gain values; converting the plurality of directional audio signals into a two- or multi-channel audio signal according to the quantity of two or more channels of a head-mounted playback device; and playing the two-channel or multi-channel audio signal through the head-mounted playback device.

Abstract: A singing scoring method and a singing scoring system based on streaming media are provided. In the singing scoring method and the singing scoring system, a first time difference between a moment at which a streaming video player starts to play a song and a moment at which an electronic device starts an audio recording program and a scoring engine is calculated. In addition, in the singing scoring method and the singing scoring system, a playing time difference of the streaming video player within every fixed period of a system time of the electronic device is continuously calculated, and the playing time difference is transferred to the scoring engine for accumulation to form a second time difference. The scoring engine then adjusts a singing time of each note in an entire musical score according to the first time difference and the second time difference.

Abstract: An active noise cancellation system for canceling noises within a predetermined bandwidth includes an analog-to-digital converter, a programmable noise-cancellation module, a first interpolation filter and a digital-to-analog converter. The analog-to-digital converter receives a first audio signal, and converts the first audio signal from an analog signal to a digital signal. The programmable noise-cancellation processes the first audio signal to generate a noise cancellation signal. The first interpolation filter receives a second audio signal and filters the second audio signal. The noise cancellation signal and the filtered second audio signal are integrated by an adder as a third audio signal, and then the digital-to-analog converter converts the third audio signal from a digital signal to an analog signal.

Abstract: The present disclosure illustrates a voice apparatus and a dual-microphone voice system with noise cancellation. A voice detector and a noise detector simultaneously receive a first voice source and a second voice source to acquire a first main signal with a voice band and a second main signal with a burst noise band respectively. A voice filter filters a voice signal of the first main signal to remain a burst noise signal with a little voice signal (i.e., a remained noise). A noise filter filters a burst noise signal of the second main signal to remain a voice signal with a little burst noise signal (i.e., a remained voice). A post-filter generates a noise reduction gain according to the remained voice and the remained noise and adjusts the remained voice according to the noise reduction gain to generate the voice signal.

Abstract: The disclosure is related to a calibration system for active noise cancellation and a speaker apparatus. The calibration system receives the signals with feedforward control or feedback control active noise cancellation. A gain adjustment element is used to adjust a gain of the signals, and a path selection switch is used to switch connection to a first operational amplifier or to a second operational amplifier. In addition to driving signals, the operational amplifier is also used to adjust a phase of the output signals. The calibration system is able to balance the gain of the signals with active noise cancellation and adjust the phase of signals of a left-channel circuit and a right-channel circuit through gain-phase adjustment. The related speaker apparatus is such as an earphone with the feedforward ANC control circuit, the feedback ANC control circuit, or a hybrid ANC circuit.

Abstract: The disclosure is related to a method and a system for self-tuning active noise canceller (STANC) and a headset apparatus. The headset apparatus is placed on a measurement device to emulate user scenario where the user receives audio signal from the headset. The STANC system receives environmental noise signal from a microphone inside the headset. The output of the STANC system acts as reverse noise signal to suppress the environmental noise signal via a speaker unit. The corresponding mixture signal is defined as an error signal. In a calibration mode, the error signal received from the measurement device can be used to update the STANC parameters. The process will be done when the error signal is lower than a predefined threshold. The final parameters can be saved as default settings for the headset apparatus in a user mode.

Abstract: A reference frequency calibration module is provided. The reference frequency calibration module includes an oscillator, a frequency divider, a phase-locked loop (PLL) and a frequency-offset calibration unit. The frequency divider couples to the oscillator. The phase-locked loop couples to the frequency divider. The frequency-offset calibration unit couples to the frequency divider and the phase-locked loop. The oscillator is configured for operatively generating an oscillating signal having an oscillating frequency. The frequency divider divides the oscillating signal having the oscillating frequency by a first division parameter to generate a first clock signal having a first reference frequency. The phase-locked loop generates a second clock signal having a second reference frequency according to the first clock signal. The frequency-offset calibration unit is configured for operatively generating the first division parameter according to the second clock signal.

Abstract: A headphone with active noise cancelling and auto-calibration method thereof is disclosed. The headphone includes a first gain-amplifier, an active noise cancelling module, a speaker, a measurement circuit, a calibration control circuit and USB driving circuit. Auto-calibration of the headphone is by use of a USB interface in the instant disclosure, so as to significantly reduce calibration time and then improve production efficiency.

Abstract: A headphone with active noise cancelling and auto-calibration method thereof is disclosed. The headphone includes a first gain-amplifier, an active noise cancelling module, a speaker, a measurement circuit, a calibration control circuit and USB driving circuit. Auto-calibration of the headphone is by means of a USB interface in the instant disclosure, so as to significantly reduce calibration time and then improve production efficiency.

Abstract: The present disclosure provides an auto-determining sampling frequency method. The method is for determining sampling frequency for an input signal of a single wire transmission interface. Each frame of the input signal includes a preamble and binary data presented in a plurality of bits. The method includes utilizing an internal sampling clock to acquire a plurality of period widths of the preamble and the binary data in the input signal and determining range of the sampling frequency according to the detected period widths of the preamble and the binary data.

Abstract: An exemplary embodiment of the present disclosure illustrates a clock rate control method. Firstly, a usage of a first input first output (FIFO) buffer in an electronic device is detected. Then, whether the usage falls within a first specific interval is determined, wherein the first specific interval has a first upper limit value and a first lower limit value. When the usage is larger than the first upper limit value, a clock rate of the inner device of the electronic device is increased; when the usage is less than the first lower limit value, the clock rate is decreased.

Abstract: The present disclosure provides an auto-determining sampling frequency method. The method is for determining sampling frequency for an input signal of a single wire transmission interface. Each frame of the input signal includes a preamble and binary data presented in a plurality of bits. The method includes utilizing an internal sampling clock to acquire a plurality of period widths of the preamble and the binary data in the input signal and determining range of the sampling frequency according to the detected period widths of the preamble and the binary data.

Abstract: This invention provides a common mode noise cancellation circuit for the unbalanced signals. The unbalanced signals come from a signal source with a first signal terminal and a second signal terminal having a first grounding potential. The common mode noise cancellation circuit comprises a grounding terminal and a subtractor. The grounding terminal with a second grounding potential is electrically coupled to the second signal terminal of the signal source through an impedance unit. The subtractor comprises a first receiving terminal, a second receiving terminal and a signal output terminal. The first receiving terminal and the second receiving terminal are electrically coupled to the first signal terminal and the second signal terminal respectively for receiving the unbalanced signals. The subtractor subtracts the noise coming from the first receiving terminal and the noise coming from the second receiving terminal to reduce the output noise of the signal output terminal.

Abstract: Disclosed herein are system, method and apparatus with environmental noise cancellation. The instant disclosure is particularly adapted to a receiver module having at least two inputs. The two inputs respectively receive a main audio portion and the audio with majority of environmental noise. The system firstly calibrates the audio signals to reduce the error caused by the difference between the two inputs. An adaptive beamforming technology and a speech extractor are respectively used to extract the environmental noise portion with less main audio and the main audio portion with less noise. After a process of time-to-frequency domain transformation, a non-linear noise suppression technology is introduced into estimating the environmental noise and acquiring a gain. After noise suppression processed with the gain, a sequence of audio signals is output after a frequency-to-time domain transformation.

Abstract: This invention provides a common mode noise cancellation circuit for the unbalanced signals. The unbalanced signals come from a signal source with a first signal terminal and a second signal terminal having a first grounding potential. The common mode noise cancellation circuit comprises a grounding terminal and a subtractor. The grounding terminal with a second grounding potential is electrically coupled to the second signal terminal of the signal source through an impedance unit. The subtractor comprises a first receiving terminal, a second receiving terminal and a signal output terminal. The first receiving terminal and the second receiving terminal are electrically coupled to the first signal terminal and the second signal terminal respectively for receiving the unbalanced signals. The subtractor subtracts the noise coming from the first receiving terminal and the noise coming from the second receiving terminal to reduce the output noise of the signal output terminal.

Abstract: A virtual signal source generating apparatus is provided. The virtual signal source generating apparatus includes at least an audio interface, a control module, a detecting module and a processing module. The control module is used for acquiring an audio signal from a signal source, wherein the audio signal includes a clock information and an audio information. The detecting module is used for detecting whether the audio interface is coupled to the signal source. The processing module is used for transmitting the audio signal. If the detecting module detects that the signal source disappears when the virtual signal source generating apparatus is in the recording mode, the control module transmits a control signal to the processing module so that the processing module transmits a virtual audio signal to a host.

Abstract: A voltage amplitude limiting circuit of a full differential circuit is provided for limiting voltage levels of a differential signal. The voltage amplitude limiting circuit includes a reference voltage generating unit and a replacing circuit. The reference voltage generating unit generates a high reference voltage and a low reference voltage. The replacing circuit is coupled to the reference voltage generating unit, a first input terminal and a second input terminal. When voltage at the first input terminal is greater than the high reference voltage, the replacing circuit uses the high reference voltage to replace the voltage at the first input terminal to serve as an output. When voltage at the first input terminal is less than the low reference voltage, the replacing circuit uses the low reference voltage to replace the voltage at the first input terminal to serve as an output.

Abstract: An exemplary embodiment of the present disclosure illustrates a clock rate control method. Firstly, a usage of a first input first output (FIFO) buffer in an electronic device is detected. Then, whether the usage falls within a first specific interval is determined, wherein the first specific interval has a first upper limit value and a first lower limit value. When the usage is larger than the first upper limit value, a clock rate of the inner device of the electronic device is increased; when the usage is less than the first lower limit value, the clock rate is decreased.

Abstract: Provided is method for automatic audio gain modulation and a related apparatus. In order to prevent the unstable signals as continuously increasing or decreasing the gain when the signals hover around an upper threshold, it's featured that the method is to predict the signal intensity as pre-adding a predict gain in a hold time before outputting signals. By determining whether the prediction exceeds a predetermined threshold or not, the suitable and adjustable gains can be obtained without exceeding the predetermined threshold. In the automatic gain control mechanism, it's to decrease the gain gradually as automatically entering an attack time, or to increase the gain gradually as performing a conventional auto-gain control procedure including entering a release time. The claimed method can modulate the gain automatically by predicting the gain, and confine the signals under a threshold. In addition to avoid sawtooth phenomena, the method can stabilize the signals.