Abstract: A device and method, respectively, obtain a first order ambisonic (FOA) signal from signals of multiple microphones, e.g., at least four or five directive microphones. The device and method determine a look direction of each microphone, and calculate a decoding matrix based on the determined look directions. The decoding matrix is a matrix suitable for decoding a FOA signal into the signals of the microphones. Further, the device and method invert the decoding matrix to obtain an encoding matrix, and encode the signals of the microphones based on the encoding matrix to obtain the FOA signal.

Abstract: A method and a device encode N audio signals, from N microphones where N?3. For each pair of the N audio signals an angle of incidence of direct sound is estimated. A-format direct sound signals are derived from the estimated angles of incidence by deriving from each estimated angle an A-format direct sound signal. Each A-format direct sound signal is a first-order virtual microphone signal, for example, a cardioids signal.

Abstract: A device and method, respectively, obtain a first order ambisonic (FOA) signal from signals of multiple microphones, e.g., at least four or five directive microphones. The device and method determine a look direction of each microphone, and calculate a decoding matrix based on the determined look directions. The decoding matrix is a matrix suitable for decoding a FOA signal into the signals of the microphones. Further, the device and method invert the decoding matrix to obtain an encoding matrix, and encode the signals of the microphones based on the encoding matrix to obtain the FOA signal.

Abstract: A method and a device encode N audio signals, from N microphones where N?3. For each pair of the N audio signals an angle of incidence of direct sound is estimated. A-format direct sound signals are derived from the estimated angles of incidence by deriving from each estimated angle an A-format direct sound signal. Each A-format direct sound signal is a first-order virtual microphone signal, for example, a cardioids signal.

Abstract: An audio signal processing apparatus, comprising a memory configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, a processor configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter configured to filter the input audio signal on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function configured to adjust a delay between the left ear transfer function and the right ear transfer function to obtain a left ear output audio signal and a right ear output audio signal.

Abstract: An audio signal processing apparatus, comprising a memory configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, a processor configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter configured to filter the input audio signal on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function configured to adjust a delay between the left ear transfer function and the right ear transfer function to obtain a left ear output audio signal and a right ear output audio signal.

Abstract: An input audio signal is separated into input audio signal components (X(k,b)). A set of two or more band branches provides output audio signal components (Y(k,b)). The set of band branches comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal and the summed audio signal(y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

Abstract: The invention relates to an audio signal processing apparatus for processing an input earpiece audio signal upon the basis of a microphone audio signal, the audio signal processing apparatus comprising a voice activity detector being configured to determine a voice activity indicator signal upon the basis of the input earpiece audio signal, a noise magnitude determiner being configured to determine a microphone noise magnitude indicator signal upon the basis of the microphone audio signal, a gain factor determiner being configured to determine a gain factor signal upon the basis of the voice activity indicator signal and the microphone noise magnitude indicator signal, and a weighter being configured to weight the input earpiece audio signal by the gain factor signal to obtain an output earpiece audio signal.

Abstract: An input audio signal (101) is separated into input audio signal components (X(k,b)). A set of two or more band branches (105e) provides output audio signal components (Y(k,b)). The set of band branches (105e) comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal (101) and the summed audio signal (y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal (103) is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

Abstract: The disclosure relates to an apparatus for manipulating an input audio signal associated to a spatial audio source within a spatial audio scenario, wherein the spatial audio source has a certain distance to a listener within the spatial audio scenario. The apparatus comprises an exciter adapted to manipulate the input audio signal to obtain an output audio signal, and a controller adapted to control parameters of the exciter for manipulating the input audio signal based on the certain distance.

Abstract: A microphone arrangement and a method using the microphone arrangement for recording surround sound in a mobile device, where the microphone arrangement comprises a first and a second microphone and arranged at a first distance to each other and configured to obtain a stereo signal, and comprises a third microphone configured to obtain a steering signal together with at least one of the first and second microphone or with a fourth microphone. The microphone arrangement also comprises a processor configured to separate the stereo signal into a front stereo signal and a back stereo signal based on the steering signal.

Abstract: An audio signal processing apparatus, comprising a memory configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, a processor configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter configured to filter the input audio signal on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function configured to adjust a delay between the left ear transfer function and the right ear transfer function to obtain a left ear output audio signal and a right ear output audio signal.

Abstract: The disclosure is based on the finding that acoustic near-field transfer functions indicating acoustic near-field propagation channels between loudspeakers and ears of a listener can be employed to pre-process audio signals. Therefore, acoustic near-field distortions of the audio signals can be mitigated. The pre-processed audio signals can be presented to the listener using a wearable frame, wherein the wearable frame comprises the loudspeakers for audio presentation. The disclosure can allow for a high quality rendering of audio signals as well as a high listening comfort for the listener. The disclosure can provide the following advantages. By means of a loudspeaker selection as a function of a spatial audio source direction, cues related to the listener's ears can be generated, making the approach more robust with regard to front/back confusion. The approach can further be extended to an arbitrary number of loudspeaker pairs.

Abstract: The invention relates to an audio signal processing apparatus (100) for processing an input earpiece audio signal (x) upon the basis of a microphone audio signal (y), the audio signal processing apparatus (100) comprising a voice activity detector (101) being configured to determine a voice activity indicator signal (xvad) upon the basis of the input earpiece audio signal (x), a noise magnitude determiner (103) being configured to determine a microphone noise magnitude indicator signal (wy) upon the basis of the microphone audio signal (y), a gain factor determiner (105) being configured to determine a gain factor signal (?G) upon the basis of the voice activity indicator signal (xvad) and the microphone noise magnitude indicator signal (wy), and a weighter (107) being configured to weight the input earpiece audio signal (x) by the gain factor signal (?G) to obtain an output earpiece audio signal.

Abstract: A microphone arrangement and a method using the microphone arrangement for recording surround sound in a mobile device, where the microphone arrangement comprises a first and a second microphone and arranged at a first distance to each other and configured to obtain a stereo signal, and comprises a third microphone configured to obtain a steering signal together with at least one of the first and second microphone or with a fourth microphone. The microphone arrangement also comprises a processor configured to separate the stereo signal into a front stereo signal and a back stereo signal based on the steering signal.

Abstract: An apparatus for computing filter coefficients for an adaptive filter is disclosed. The adaptive filter is used for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal. The apparatus has: an echo decay modeling means for modeling a decay behavior of an acoustic environment and for providing a corresponding echo decay parameter; and computing means for computing the filter coefficients of the adaptive filter on the basis of the echo decay parameter. A corresponding method has: providing echo decay parameters determined by means of an echo decay modeling means; and computing the filter coefficients of the adaptive filter on the basis of the echo decay parameters.

Abstract: The disclosure relates to an apparatus for manipulating an input audio signal associated to a spatial audio source within a spatial audio scenario, wherein the spatial audio source has a certain distance to a listener within the spatial audio scenario. The apparatus comprises an exciter adapted to manipulate the input audio signal to obtain an output audio signal, and a controller adapted to control parameters of the exciter for manipulating the input audio signal based on the certain distance.

Abstract: The disclosure is based on the finding that acoustic near-field transfer functions indicating acoustic near-field propagation channels between loudspeakers and ears of a listener can be employed to pre-process audio signals. Therefore, acoustic near-field distortions of the audio signals can be mitigated. The pre-processed audio signals can be presented to the listener using a wearable frame, wherein the wearable frame comprises the loudspeakers for audio presentation. The disclosure can allow for a high quality rendering of audio signals as well as a high listening comfort for the listener. The disclosure can provide the following advantages. By means of a loudspeaker selection as a function of a spatial audio source direction, cues related to the listener's ears can be generated, making the approach more robust with regard to front/back confusion. The approach can further be extended to an arbitrary number of loudspeaker pairs.

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; receiving user gain input; generating a linear gain factor and a non-linear gain factor using the user gain input; modifying the non-linear gain factor using absolute threshold of hearing and power of the input signal to generate a modified non-linear gain factor; and, applying the linear gain factor and the modified non-linear gain factor to the audio signal.