Methods, apparatus, systems and articles of manufacture are disclosed for audio watermarking and, more particularly, methods and apparatus to inspect characteristics of multichannel audio. An example apparatus disclosed herein includes an audio demultiplexer to obtain a first audio subchannel and a second audio subchannel of a multichannel audio signal and a watermark detector to detect watermarks in the first audio subchannel and watermarks in the second audio subchannel. The example apparatus further includes a channel characteristic inspector to compare a number of watermarks detected in the first audio subchannel and a number of watermarks detected in the second audio subchannel to determine whether the first and second audio subchannels were watermarked in accordance with a watermark encoder configuration and a result alerter to distribute an alert in response to a determination that the first and second audio subchannels were not watermarked in accordance with the watermark encoder configuration.

Systems, devices, and methods are described for reducing degradation of a voice recognition input. An always listening device may always be listening for voice commands via a microphone and may experience interference from unwanted audio such as from the output audio of television speakers. The always listening device may receive data associated with the output audio over a first communications channel. The always listening device may also receive, on a second communications channel, timing information associated with data. The always listening device may adjust admission of the audio received by the microphone to enable it to arrive at approximately the same time as the data received via the first communications channel. The unwanted output audio included in the audio received via the microphone may then be determined and may be removed so that a voice command in the audio received by the microphone may be processed.

Systems, devices, and methods are described for reducing degradation of a voice recognition input. An always listening device may always be listening for voice commands via a microphone and may experience interference from unwanted audio such as from the output audio of television speakers. The always listening device may receive data associated with the output audio over a first communications channel. The always listening device may also receive, on a second communications channel, timing information associated with data. The always listening device may adjust admission of the audio received by the microphone to enable it to arrive at approximately the same time as the data received via the first communications channel. The unwanted output audio included in the audio received via the microphone may then be determined and may be removed so that a voice command in the audio received by the microphone may be processed.

The present disclosure provides method and system for voice separation based on DUET algorithm, and the method comprises receiving signals from microphones; performing a Fourier transform on the received signals; calculating a relative attenuation parameter and a relative delay parameter for each data point; selecting a clustering range for the relative delay parameters based on a distance between the microphones and a sampling frequency of the microphones, clustering the data points within the clustering range for the relative delay parameters into subsets, and performing an inverse Fourier transform on each subsets. According to the present disclosure, it is possible to provide an efficient and intelligent solution to deploy DUET on the software and/or hardware.

The present disclosure provides method and system for voice separation based on DUET algorithm, and the method comprises receiving signals from microphones; performing a Fourier transform on the received signals; calculating a relative attenuation parameter and a relative delay parameter for each data point; selecting a clustering range for the relative delay parameters based on a distance between the microphones and a sampling frequency of the microphones, clustering the data points within the clustering range for the relative delay parameters into subsets, and performing an inverse Fourier transform on each subsets. According to the present disclosure, it is possible to provide an efficient and intelligent solution to deploy DUET on the software and/or hardware.

Disclosed herein are new techniques carried out by a computing system for determining delays of various components of an audio system to allow for accurate correction of these delays, which may improve the audio quality of live performances for listeners who hear audio reproduced by loudspeakers at live performance venues. In one implementation the computing system, which may comprise a transmitter device and one or more receiver devices, may be configured to perform functions, including receiving a first audio signal, receiving, via an audio input interface of the receiver, a second audio signal, and determining, based on the first audio signal and the second audio signal, an audio delay that is associated with the second audio signal. The computing system may be configured to perform further functions, including based on a determined cross-correlation between a downsampled audio signal and a filtered second audio signal, determining the audio signal delay.

Disclosed herein are new techniques carried out by a computing system for determining delays of various components of an audio system to allow for accurate correction of these delays, which may improve the audio quality of live performances for listeners who hear audio reproduced by loudspeakers at live performance venues. In one implementation the computing system, which may comprise a transmitter device and one or more receiver devices, may be configured to perform functions, including receiving a first audio signal, receiving, via an audio input interface of the receiver, a second audio signal, and determining, based on the first audio signal and the second audio signal, an audio delay that is associated with the second audio signal. The computing system may be configured to perform further functions, including based on a determined cross-correlation between a downsampled audio signal and a filtered second audio signal, determining the audio signal delay.

A sampling rate processing method performed by a computer device are disclosed. The method includes: obtaining a first audio signal recorded by a transmitting device, the first audio signal being recorded according to an initial sampling rate of the transmitting device; obtaining a second audio signal recorded by a receiving device during playing of the first audio signal, the second audio signal being recorded according to the initial sampling rate; determining a frequency response gain value of the receiving device according to a power spectrum of the first audio signal and a power spectrum of the second audio signal; determining a target sampling rate of the transmitting device according to the initial sampling rate and the frequency response gain value; and configuring the transmitting device to record audio signals according to the target sampling rate.

A sampling rate processing method performed by a computer device are disclosed. The method includes: obtaining a first audio signal recorded by a transmitting device, the first audio signal being recorded according to an initial sampling rate of the transmitting device; obtaining a second audio signal recorded by a receiving device during playing of the first audio signal, the second audio signal being recorded according to the initial sampling rate; determining a frequency response gain value of the receiving device according to a power spectrum of the first audio signal and a power spectrum of the second audio signal; determining a target sampling rate of the transmitting device according to the initial sampling rate and the frequency response gain value; and configuring the transmitting device to record audio signals according to the target sampling rate.

A system is configured to manage the recording of audio content between multiple parties participating in a two way audio communication over a plurality of user devices. The audio files recorded during the two way communication can be saved locally and then uploaded to a management server in real time or at a later time. Pluralities of single party audio datasets are organized to provide a synchronized timeline. An editor interface is displayed to an editor user that allows synchronized timelines of single party datasets to be viewed and searched in transcribed form. By selecting words from a transcript users may remove corresponding spoken audio from single party audio datasets before they are merged into a multi-party audio dataset.