Examples of methods for audio session classification are described herein. In some examples, a method may include determining, at a first classification stage, whether an audio session is classifiable with predetermined criteria. In some examples, the method may include classifying, at a second classification stage, the audio session based on a machine learning analysis of metadata in a case that the audio session is not classifiable at the first classification stage.

An apparatus for generating one or more audio output channels is provided. The apparatus includes a parameter processor for calculating output channel mixing information and a downmix processor for generating the one or more audio output channels. The downmix processor is configured to receive an audio transport signal including one or more audio transport channels, wherein two or more audio object signals are mixed within the audio transport signal, and wherein the number of the one or more audio transport channels is smaller than the number of the two or more audio object signals. The audio transport signal depends on a first mixing rule and on a second mixing rule. The first mixing rule indicates how to mix the two or more audio object signals to obtain a plurality of premixed channels. Moreover, the second mixing rule indicates how to mix the plurality of premixed channels.

A method and system dynamically adjusts the audio of an audio and video signal to improve its overall sound quality and dialog intelligibility. Some embodiments use gain, equalization, audio signal compression and spatial enhancement (reverb) on individual channels of a multichannel audio signal.

Systems and methods for adjusting bass levels of a multi-channel audio signal include, among other features, (i) receiving the multi-channel signal via a playback device; (ii) separating, from the multi-channel signal, low-frequency signals comprising frequencies less than a threshold frequency; (iii) determining electrical energies of the low-frequency signals; (iv) determining a first energy by summing the electrical energies of the low-frequency signals; (v) consolidating the low-frequency signals into a consolidated low-frequency signal; (vi) determining a second energy by determining an electrical energy of the consolidated low-frequency signal; (vii) generating a gain-adjusted low-frequency signal by adjusting a gain of the consolidated low-frequency signal based on both (a) the first energy and (b) the second energy; (viii) generating a gain-adjusted multi-channel signal by mixing the gain-adjusted low-frequency signal back into the multi-channel signal; and (ix) using the gain-adjusted multi-channel signal to play back gain-adjusted multi-channel audio content via the playback device.

A display device according to an embodiment of the present disclosure is to provide a display device for automatically configuring an audio channel suitable for a location of each of a plurality of external speakers when the plurality of external speakers are connected, and an operating method thereof, which establish each of the plurality of external speakers and transmit audio signals to the plurality of external speakers according to the established roles.

In general, techniques are described by which to support scalable unified audio rendering. A device comprising an audio decoder, a memory, and a processor may be configured to perform various aspects of the techniques. The audio decoder may decode, from a bitstream, first audio data and second audio data. The memory may store the first audio data and the second audio data. The processor may render the first audio data into first spatial domain audio data for playback by virtual speakers at a set of virtual speaker locations, and render the second audio data into second spatial domain audio data for playback by the virtual speakers at the set of virtual speaker locations. The processor may also mix the first spatial domain audio data and the second spatial domain audio data to obtain mixed spatial domain audio data, and convert the mixed spatial domain audio data to scene-based audio data.

An circuit includes: a plurality of analog-to-digital converters (ADCs) and a control chip. The control chip is utilized for instructing a target ADC to output audio data of a target channel during a target period, and utilized for instructing remaining ADCs not to output audio data in the target period. Then, the control chip defines data timing of the target channel and other channels based on the data receiving time point of the audio data of the target channel. The plurality of ADCs would process analog audio signals of a plurality of channels and output audio data of the plurality of channels according to an assigned order configured by the control chip to form a serial data signal. The control chip separates the audio data of different channels from the serial data signal according to the data timing of the plurality of channels.

Systems and methods for adjusting bass levels of a multi-channel audio signal include, among other features, (i) receiving the multi-channel signal via a playback device; (ii) separating, from the multi-channel signal, low-frequency signals comprising frequencies less than a threshold frequency; (iii) determining electrical energies of the low-frequency signals; (iv) determining a first energy by summing the electrical energies of the low-frequency signals; (v) consolidating the low-frequency signals into a consolidated low-frequency signal; (vi) determining a second energy by determining an electrical energy of the consolidated low-frequency signal; (vii) generating a gain-adjusted low-frequency signal by adjusting a gain of the consolidated low-frequency signal based on both (a) the first energy and (b) the second energy; (viii) generating a gain-adjusted multi-channel signal by mixing the gain-adjusted low-frequency signal back into the multi-channel signal; and (ix) using the gain-adjusted multi-channel signal to play back gain-adjusted multi-channel audio content via the playback device.

The present invention relates to methods for generating a conversion filter (KF) for converting a multidimensional original audio signal (AA) into a two-dimensional listening audio signal (HA), comprising the following steps: 1. Transformation of a time-based original audio signal (PAA) into a frequency-based original audio signal (FAA) 2. Sequential optimization of a basis conversion matrix (BKM) for converting the frequency-based original audio signal (FAA) into a frequency-based listening audio signal (FHA) using a first optimization algorithm (KA1), preferably starting from low frequencies and ascending at least up to a switching frequency (UF) 3. Sequential optimization of the basis conversion matrix (BKA) for converting the frequency-based original audio signal (FAA) into a frequency-based listening audio signal (FHA) using a second optimization algorithm (KA2), preferably starting from the switching frequency (UF) and ascending to high frequencies 4. Storing the optimized basis conversion matrix (BKA) of the correlation between the frequency-based original audio signal (FAA) and the frequency-based listening audio signal (FHA) in a frequency-based conversion matrix (FKM) 5. Transforming the frequency-based conversion matrix (FKM) into a time-based conversion matrix (PKM) as a conversion filter (KF).

An apparatus includes one or more processors configured to receive orientation data and to select, based on the orientation data, a particular filter from among multiple filters. The one or more processors are configured to perform signal processing operations associated with three-dimensional (3D) sound data based on the particular filter.