摘要:
A method and a system for correcting energy distributions of audio signal are proposed. The method is applicable to a head-mounted device having a motion sensor, a left speaker, and a right speaker and includes the following steps. A rotation angle of the head-mounted device is detected by the motion sensor. Dual-channel audio signals corresponding to the left and right speakers are obtained. The dual-channel audio signals are converted to multi-channel audio signals with the number of channels greater than or equal to 5. Four acoustic source positions of the left and right speakers are defined to convert the multi-channel audio signals to four-channel audio signals of the left and right speakers. Energy distributions of the four-channel audio signals of the left and right speakers are corrected according to the rotation angle and the four acoustic source positions to respectively generate a left output signal and a right output signal.
摘要:
A voice signal processing apparatus and a voice signal processing method are provided. Calculate a value of an interpolation parametric function corresponding to a sampling signal frame according to three consecutive sample values in the sampling signal frame, and calculate an interpolated value between two adjacent sampling points in a frequency-lowered signal frame according to the value of the interpolation parametric function.
摘要:
A speech recognition apparatus and a speech recognition method are provided. In the invention, whether an original voice sampling signal corresponding to a target voice frame is a consonant signal is determined according to at least one of a ratio of an energy of a low-pass sampling signal to an energy of the original voice sampling signal and a ratio value of an energy of a second consonant frequency band signal.
摘要:
In a device including a headset jack having an external microphone contact, and an external microphone signal path connected to the external microphone contact, a method including detecting whether a headset plug is plugged into the headset jack and, if the headset plug is detected, selecting an audio signal received from the external microphone signal path, recording the selected audio signal, to produce a recorded audio signal, and determining if an external microphone is connected to the external microphone contact based on the recorded audio signal.
摘要:
A method and an apparatus for audio signal processing evaluation are provided. The audio signal processing is performed on a synthesized audio signal to generate a processed audio signal. The synthesized audio signal is generated by adding a secondary signal into a master signal. The master signal is merely a speech signal. The signal processing is related to removing the secondary signal from the synthesized audio signal. The sound characteristics of the processed audio signal and the master signal are obtained, respectively. The sound characteristics include text content, and the text content is generated by performing speech-to-text on the processed audio signal and the master signal. The audio signal processing is evaluated according to the compared result between the sound characteristics of the processed audio signal and the master signal. The compared result includes the correctness of the text content of the processed audio signal relative to the master signal.
摘要:
A Finite Impulse Response (FIR) filter is configured to minimize delay and maximize passband power by adjusting the filter coefficients applied to the sampled values. The FIR filter obtains an input signal and samples the input signal to generate a set of sampled input values. The FIR filter generates a set of filter coefficients, with each filter coefficient based on a corresponding sampled input value in the set of sample input values. The FIR filter selects a subset of sampled input values that have been most recently sampled from the input signal, and selects a subset of filter coefficients corresponding to sampled input values that are not the most recently sampled. The subset of sampled input values is combined with the subset of filter coefficients to generate an output value for the FIR filter.
摘要:
A specific sound source automatic adjusting method and an electronic device using the same are provided. The electronic device includes a first audio recognition unit, a first multi-sound source determination unit, a directivity analysis unit, a directional separation unit, a second audio recognition unit, a second multi-sound source determination unit and an audio adjustment unit. If the number of sound sources of the original sound signal is larger than or equal to 2, the directivity analysis unit performs a directionality analysis procedure on the original sound signal. The directional separation unit separates out at least one specific directional sub-signal from the original sound signal according to the result of directional analysis procedure. If the number of sound sources of the specific directional sub-signal is equal to 1, the audio adjustment unit performs a sound source adjustment procedure.
摘要:
A sound outputting device, a processing device and a sound controlling method thereof are provided. The sound controlling method includes the following steps. An original left sound signal and an original right sound signal are received. The original left sound signal and the original right sound signal are transformed to be a virtual left sound signal and a virtual right sound signal of a virtual sound source. A rotation degree of a user is detected. The virtual left sound signal and the virtual right sound signal are transformed to be an updated left sound signal and an updated right sound signal.
摘要:
An adjusting system and an adjusting method for equalization processing are provided. Frequency band energies of sound receiving signals are obtained. The frequency band energies correspond to different frequency bands, respectively. Target gains corresponding to frequency bands are determined according to the frequency band energies. Frequency responses of filtering processing with respect to a plurality of center frequencies are obtained. Equalization gains corresponding to the frequency bands and having the least gain error are determined. The gain error is related to a difference between the amplitude obtained after the equalization gains are reflected on the frequency responses corresponding to the filtering processing and the target gains. The equalization gains are inputted into the filtering processing according to the corresponding frequency bands. Accordingly, the impact of the filtering processing can be reduced.
摘要:
An adjusting system and an adjusting method for equalization processing are provided. Frequency band energies of sound receiving signals are obtained. The frequency band energies correspond to different frequency bands, respectively. Target gains corresponding to frequency bands are determined according to the frequency band energies. Frequency responses of filtering processing with respect to a plurality of center frequencies are obtained. Equalization gains corresponding to the frequency bands and having the least gain error are determined. The gain error is related to a difference between the amplitude obtained after the equalization gains are reflected on the frequency responses corresponding to the filtering processing and the target gains. The equalization gains are inputted into the filtering processing according to the corresponding frequency bands. Accordingly, the impact of the filtering processing can be reduced.