In some implementations, a user device may receive input that triggers transmission of information via sound. The user device may select an audio clip based on a setting associated with the device, and may modify a digital representation of the selected audio clip using an encoding algorithm and based on data associated with a user of the device. The user device may transmit, to a remote server, an indication of the selected audio clip, an indication of the encoding algorithm, and the data associated with the user. The user device may use a speaker to play audio, based on the modified digital representation, for recording by other devices. Accordingly, the user device may receive, from the remote server and based on the speaker playing the audio, a confirmation that users associated with the other devices have performed an action based on the data associated with the user of the device.

A device includes an encoder configured to generate a first high-band portion of a first signal based on a left signal and a right signal. The encoder is also configured to generate a set of adjustment gain parameters based on a high-band non-reference signal and a synthesized signal. The high-band non-reference signal corresponds to one of a left high-band portion of the left signal or a right high-band portion of the right signal.

A display apparatus and a method for processing audio are provided, the display apparatus includes a circuit board provided with a hybrid circuit, a filter circuit and a speaker; the hybrid circuit is configured to receive an original audio signal and superpose a first sub-signal of the original audio signal on a second sub-signal of the original audio signal to obtain a hybrid audio signal; the first sub-signal includes at least one channel of audio signal, the second sub-signal includes at least two channels of audio signal; the filter circuit is configured to filter the hybrid audio signal according to a frequency characteristic of the first sub-signal and the second sub-signal to obtain a restored original audio signal; and the speaker, connected with the filter circuit, is configured to output the restored original audio signal.

Computer-implemented methods, computer program products, and computer systems for mitigating frequency loss may include one or more processors configured for receiving first audio data corresponding to unobstructed user utterances, receiving second audio data corresponding to first obstructed user utterances, generating a frequency loss (FL) model representing frequency loss between the first audio data and the second audio data, receiving third audio data corresponding to one or more second obstructed user utterances, processing the third audio data using the FL model to generate fourth audio data corresponding to a frequency loss mitigated version of the second obstructed user utterances, and transmitting the fourth audio data to a recipient computing device. The first obstructed user utterances are obstructed by a facemask and the one or more second obstructed user utterances is obstructed by the facemask. The FL model may be executed as an audio plugin in a web conferencing program.

The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank (501) comprising an analysis transformation unit (601) having a frequency resolution of Δf; and an analysis window (611) having a duration of D.sub.A; the analysis filter bank (501) being configured to provide a set of analysis subband signals from the low frequency component of the signal; a nonlinear processing unit (502, 650) configured to determine a set of synthesis subband signals based on a portion of the set of analysis subband signals, wherein the portion of the set of analysis subband signals is phase shifted by a transposition order T; and a synthesis filter bank (504) comprising a synthesis transformation unit (602) having a frequency resolution of QΔf; and a synthesis window (612) having a duration of D.sub.S; the synthesis filter bank (504) being configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein Q is a frequency resolution factor with Q≥1 and smaller than the transposition order T; and wherein the value of the product of the frequency resolution Δf and the duration D.sub.A of the analysis filter bank is selected based on the frequency resolution factor Q.

An encoder for encoding a parametric spectral representation (f) of auto-regressive coefficients that partially represent an audio signal. The encoder includes a low-frequency encoder configured to quantize elements of a part of the parametric spectral representation that correspond to a low-frequency part of the audio signal. It also includes a high-frequency encoder configured to encode a high-frequency part (f.sup.H) of the parametric spectral representation (f) by weighted averaging based on the quantized elements ({circumflex over (f)}.sup.L) flipped around a quantized mirroring frequency ({circumflex over (f)}.sub.m), which separates the low-frequency part from the high-frequency part, and a frequency grid determined from a frequency grid codebook in a closed-loop search procedure. Described are also a corresponding decoder, corresponding encoding/decoding methods and UEs including such an encoder/decoder.

Disclosed is a method of processing a residual signal for audio coding and an audio coding apparatus. The method learns a feature map of a reference signal through a residual signal learning engine including a convolutional layer and a neural network and performs learning based on a result obtained by mapping a node of an output layer of the neural network and a quantization level of index of the residual signal.

A method comprising: obtaining a first audio direction parameter value for each sub-band of a sub-frame of a frame of an audio signal; obtaining a second audio direction parameter value for the sub-frame of the frame of the audio signal for one or more audio objects associated with said audio signal; and determining a bit-efficient encoding for each first audio direction parameter value of the sub-frame based on a similarity between the first audio direction parameter value for each sub-band and the second audio direction parameter values for the one or more audio objects.

Methods and apparatus are disclosed for supplementing partially readable and/or inaccurate codes. An example apparatus includes a watermark analyzer to select a first watermark and a second watermark decoded from media; a comparator to compare a first decoded timestamp of the first watermark to a second decoded timestamp of the second watermark; and a timestamp adjuster to adjust the second decoded timestamp based on the first decoded timestamp of the second watermark when at least a threshold number of symbols of the second decoded timestamp match corresponding symbols of the first decoded timestamp.

The present disclosure discloses an audio encoding and decoding method and an audio encoder and decoder. The audio encoding method includes: obtaining a current frame of an audio signal, where the current frame includes a high frequency band signal and a low frequency band signal; obtaining a first encoding parameter based on the high frequency band signal and the low frequency band signal; obtaining a second encoding parameter of the current frame based on the high frequency band signal, where the second encoding parameter includes tone component information; and performing bitstream multiplexing on the first encoding parameter and the second encoding parameter, to obtain an encoded bitstream.