The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L.sub.a, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length L.sub.s, generating a frame of the output signal.

This application relates to the field of signal processing technologies and headsets, and provides a speech signal processing method and apparatus, to provide a full-band low-noise speech signal. The method is applied to a headset including at least two speech collectors, where the at least two speech collectors include an ear canal speech collector and at least one external speech collector. The method includes: preprocessing a speech signal that is in a first frequency band and that is collected by the ear canal speech collector, to obtain a first speech signal; preprocessing a speech signal that is in a second frequency band and that is collected by the at least one external speech collector, to obtain an external speech signal, where frequency ranges of the first frequency band and the second frequency band are different; performing correlation processing on the first speech signal and the external speech signal to obtain a second speech signal; and outputting a target speech signal, where the target speech signal includes the first speech signal and the second speech signal.

This application relates to the field of signal processing technologies and headsets, and provides a speech signal processing method and apparatus, to provide a full-band low-noise speech signal. The method is applied to a headset including at least two speech collectors, where the at least two speech collectors include an ear canal speech collector and at least one external speech collector. The method includes: preprocessing a speech signal that is in a first frequency band and that is collected by the ear canal speech collector, to obtain a first speech signal; preprocessing a speech signal that is in a second frequency band and that is collected by the at least one external speech collector, to obtain an external speech signal, where frequency ranges of the first frequency band and the second frequency band are different; performing correlation processing on the first speech signal and the external speech signal to obtain a second speech signal; and outputting a target speech signal, where the target speech signal includes the first speech signal and the second speech signal.

A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

An audio transmitter processor for generating an error protected frame using encoded audio data of an audio frame, the encoded audio data for the audio frame having a first amount of information units and a second amount of information units, has: a frame builder for building a codeword frame having a codeword raster, wherein the frame builder is configured to determine a border between a first amount of information units and a second amount of information units so that a starting information unit of the second amount of information units coincides with a codeword border; and an error protection coder to obtain a plurality of processed codewords representing the error protected frame.

A method including receiving a plurality of audio channels based on an audio stream, applying a model based on at least one acoustic perception algorithm to the plurality of audio channels to generate a first modelled audio stream, quantizing the plurality of audio channels using a first set of quantization parameters, dequantizing the quantized plurality of audio channels using the first set of quantization parameters, applying the model based on at least one acoustic perception algorithm to the dequantized plurality of audio channels to generate a second modelled audio stream, comparing the first modelled audio stream and the second modelled audio stream, in response to determining the comparison of the first modelled audio stream and the second modelled audio stream does not meet a criterion, generating a second set of quantization parameters, and quantizing the plurality of audio channels using the second set of quantization parameters.

A method of operating an encoder or a decoder. The method comprises receiving an analysis signal of an audio signal and a filtered analysis signal, combining the filtered signal with the analysis signal to generate a combined signal using a maximum function that provides at least one of a maximum positive value at each index i of the combined signal and a maximum negative value at each index i of the combined signal. The method comprises identifying broad peaks and narrow peaks of the combined signal.

A method of operating an encoder or a decoder. The method comprises receiving an analysis signal of an audio signal and a filtered analysis signal, combining the filtered signal with the analysis signal to generate a combined signal using a maximum function that provides at least one of a maximum positive value at each index i of the combined signal and a maximum negative value at each index i of the combined signal. The method comprises identifying broad peaks and narrow peaks of the combined signal.

The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L.sub.a, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length L.sub.s, generating a frame of the output signal.