Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

Techniques of conducting an online meeting involve outputting ambient sound to a participant of an online meeting. Along these lines, in an online meeting during which a participant wears headphones, the participant's computer receives microphone input that contains both speech from the participant and ambient sound that the participant may wish to hear. In response to receiving the microphone input, the participant's computer separates low-volume sounds from high-volume sounds. However, instead of suppressing this low-volume sound from the microphone input, the participant's computer renders this low-volume sound. In most cases, this low-volume sound represents ambient sound generated in the vicinity of the meeting participant. The participant's computer then mixes the low-volume sound with speech received from other conference participants to form output in such a way that the participant may distinguish this sound from the received speech. The participant's computer then provides the output to the participant's headphones.

In a virtual sound image control system according to the present invention, a signal processor generates the acoustic signal and outputs the acoustic signal to the two-channel loudspeakers so as to create a virtual sound image to be perceived by a user as a stereophonic sound image. The two-channel loudspeakers are arranged such that a first listening area and a second listening area for the user are symmetric to each other with respect to a virtual plane including a virtual line segment connecting the two-channel loudspeakers together.

A signal processing device includes phase rotation units which rotate a phase of a signal A and generate two signals having a phase difference of θ, and a control unit which performs transition of θ over time. The control unit controls phases so that θ is approximately 0 degrees at a time point T0 and θ is approximately 180 degrees at a time point T1.

The disclosure relates to an audio signal processing apparatus comprising a determiner being configured to determine a filter matrix C on the basis of an acoustic transfer function matrix H and a target acoustic transfer function matrix VH, wherein the acoustic transfer function matrix H comprises transfer functions of acoustic propagation paths between loudspeakers and a listener and the target acoustic transfer function matrix VH comprises target transfer functions of target acoustic propagation paths, wherein the target acoustic propagation paths are defined by a target arrangement of virtual loudspeaker positions relative to the listener, a filter being configured to filter the input audio signal on the basis of the filter matrix C to obtain filtered input audio signals, and a combiner being configured to combine the filtered input audio signals to obtain output audio signals.

An apparatus and method of blind detection of binauralized audio. If the input content is detected as binaural, a second binauralization may be avoided. In this manner, the user experience avoids audio artifacts introduced by multiple binauralizations.

A sound signal processing device includes: a vocal remover which generates a first output signal based on first-channel and second-channel sound signals and a first coefficient indicating a vocal bandwidth to be removed; a surround sound processor which generates a second output signal by adding a surround sound effect to the first output signal; an amplifier which amplifies a signal at an amplification factor that is based on a second coefficient; a synthesizer which synthesizes the second output signal with one of the first-channel and second-channel sound signals, and synthesizes a signal that is the second output signal inverted with another one of the first-channel and second-channel sound signals; and a coefficient determination unit which sets the second coefficient such that the amplification factor, used when the vocal bandwidth to be removed is greater than a first bandwidth, is greater than the amplification factor for the first bandwidth.

Methods for dialogue enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation, applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; and combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.

Systems and methods for stereo separation and directional suppression are provided. An example method includes receiving a first audio signal, representing sound captured by a first microphone associated with a first location, and a second audio signal, representing sound captured by a second microphone associated with a second location. The microphones comprise omni-directional microphones. The distance between the first and second microphones is limited by the size of a mobile device. A first channel signal of a stereo signal is generated by forming, based on the first and second audio signals, a first beam at the first location. A second channel signal of the stereo signal is generated by forming, based on the first and second audio signals, a second beam at the second location. First and second directions, associated respectively with the first and second beams, are fixed relative to a line between the first and second locations.