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.

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.

An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal. Further, the comparator determines a comparison parameter from the plurality of comparison parameters, wherein the determined comparison parameter fulfils a predefined criterion.

Described herein are techniques, devices, and systems for selectively using a music-capable audio codec on-demand during a communication session. A user equipment (UE) may adaptively transition between using a first audio codec that provides a first audio bandwidth and a second audio codec (e.g., the EVS-FB codec) that provides a second audio bandwidth that is greater than the first audio bandwidth. The transition to the second audio codec may occur in response to determining that sound in the environment of the UE includes frequencies outside of a range of frequencies associated with a human voice, such as by determining that music is being played in the environment of the UE, which allows for selectively using a music-capable audio codec when it would be beneficial to do so.

A method, a computer program product, an encoder and a monitoring device for encoding an audio signal with variable bitrate, wherein: an audio signal comprising a plurality of successive audio frames is received; and for each successive audio frame of the audio signal: the audio frame is represented in a frequency domain with respect to a plurality of frequency sub-bands; the audio frame is classified in each frequency sub-band as either background or foreground using a background model specific to the frequency sub-band; each successive audio frame of the audio signal is encoded, wherein a number of bits is allocated for each frequency sub-band of the audio frame, wherein the number of bits allocated for a frequency sub-band is higher if the audio frame is classified as foreground in the frequency sub-band than if the audio frame is classified as background in the frequency sub-band.

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Disclosed are systems and methods to modify the Bluetooth mono HFP protocol to support bi-directional stereo operation for high bandwidth audio including 12-KHz wide-band, 16-KHz super wide-band (SWB), and 24-KHz full band (FB) audio. The techniques leverage the larger packet size and longer duty cycle of the 2-EV5 transport packet and expand the block size of the audio frames generated by the AAC-ELD codec to increase the maximum data throughput from the 64 kbps of the mono HFP protocol to 192 kbps using a stereo HFP protocol. The increased throughput not only supports stereo operations, but allows the transport of redundant or FEC packets for increased robustness against packet loss. In one aspect, the AAC-ELD codec may be configured for dynamic bit rate switching to flexibly perform trade-offs between audio quality and robustness against packet loss. The stereo HFP may configure the maximum throughput based on the desired audio quality.

Disclosed are systems and methods to modify the Bluetooth mono HFP protocol to support bi-directional stereo operation for high bandwidth audio including 12-KHz wide-band, 16-KHz super wide-band (SWB), and 24-KHz full band (FB) audio. The techniques leverage the larger packet size and longer duty cycle of the 2-EV5 transport packet and expand the block size of the audio frames generated by the AAC-ELD codec to increase the maximum data throughput from the 64 kbps of the mono HFP protocol to 192 kbps using a stereo HFP protocol. The increased throughput not only supports stereo operations, but allows the transport of redundant or FEC packets for increased robustness against packet loss. In one aspect, the AAC-ELD codec may be configured for dynamic bit rate switching to flexibly perform trade-offs between audio quality and robustness against packet loss. The stereo HFP may configure the maximum throughput based on the desired audio quality.

Aspects of the disclosure provide methods and apparatuses (e.g., client devices and server devices) for audio processing. In some examples, a client device includes processing circuitry. The processing circuitry transmits, to a server device, a selection signal indicative of an audio encoding configuration for encoding audio content in an audio input. The processing circuitry receives, from the server device, an encoded bitstream in response to the transmitting of the selection signal. The encoded bitstream includes the audio content that is encoded according to the audio encoding configuration. The processing circuitry renders audio signals based on the encoded bitstream.

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.