Abstract: A method for acoustic scene playback is described, which comprises: providing recording data comprising microphone signals of microphone setups positioned within an acoustic scene and microphone metadata of the microphone setups, each of the microphone setups has a recording spot which is a center position of the respective microphone setup; specifying a virtual listening position within the acoustic scene; assigning each microphone setup Virtual Loudspeaker Objects, VLOs, wherein each VLO is an abstract sound output object within a virtual free field; generating an encoded data stream based on the recording data, the virtual listening position and VLO parameters of the VLOs assigned to the microphone setups; and decoding the encoded data stream based on a playback setup, thereby generating a decoded data stream; and feeding the decoded data stream to a rendering device, thereby driving the rendering device to reproduce sound of the acoustic scene at the virtual listening position.

Abstract: A sound signal processing apparatus including a plurality of microphones, where each microphone is configured to receive the sound signal from the target source, a processor configured to estimate a first power measure on the basis of the sound signal from the target source received by a first microphone of the microphones and a second power measure on the basis of the sound signal from the target source received by at least a second microphone of the microphones, which is located more distant from the target source than the first microphone, and the processor is further configured to determine a gain factor on the basis of a ratio between the second power measure and the first power measure, and an amplifier configured to apply the gain factor to the sound signal from the target source received by the first microphone.

Abstract: A method for acoustic scene playback is described, which comprises: providing recording data comprising microphone signals of microphone setups positioned within an acoustic scene and microphone metadata of the microphone setups, each of the microphone setups has a recording spot which is a center position of the respective microphone setup; specifying a virtual listening position within the acoustic scene; assigning each microphone setup Virtual Loudspeaker Objects, VLOs, wherein each VLO is an abstract sound output object within a virtual free field; generating an encoded data stream based on the recording data, the virtual listening position and VLO parameters of the VLOs assigned to the microphone setups; and decoding the encoded data stream based on a playback setup, thereby generating a decoded data stream; and feeding the decoded data stream to a rendering device, thereby driving the rendering device to reproduce sound of the acoustic scene at the virtual listening position.

Abstract: A sound signal processing apparatus including a plurality of microphones, where each microphone is configured to receive the sound signal from the target source, a processor configured to estimate a first power measure on the basis of the sound signal from the target source received by a first microphone of the microphones and a second power measure on the basis of the sound signal from the target source received by at least a second microphone of the microphones, which is located more distant from the target source than the first microphone, and the processor is further configured to determine a gain factor on the basis of a ratio between the second power measure and the first power measure, and an amplifier configured to apply the gain factor to the sound signal from the target source received by the first microphone.

Abstract: The disclosed active noise suppression headphone system is directed to a headphone system that is capable of substantially suppressing high or low frequency interfering noise that penetrate through a headphone earpiece from multiple directions. An external microphone mounted with a housing of a headphone earpiece senses ambient noise outside of the earpiece. The sensed ambient noise may be processed through at least one parallel filter bank arranged in at least one headphone earpiece. Each parallel filter bank may include adaptively linked filters. The output of these filters may be amplified based on weighting factors that are dependent upon the sensed ambient noise and that are generated by a filtered x least mean square circuit. The amplified filtered outputs may be summed to generate an antinoise signal that is in input to a loudspeaker within the headphone earpiece that substantial suppresses the ambient noise before it can be perceived by an end user of the headphones.

Abstract: The disclosed active noise suppression headphone system is directed to a headphone system that is capable of substantially suppressing high or low frequency interfering noise that penetrate through a headphone earpiece from multiple directions. An external microphone mounted with a housing of a headphone earpiece senses ambient noise outside of the earpiece. The sensed ambient noise may be processed through at least one parallel filter bank arranged in at least one headphone earpiece. Each parallel filter bank may include adaptively linked filters. The output of these filters may be amplified based on weighting factors that are dependent upon the sensed ambient noise and that are generated by a filtered x least mean square circuit. The amplified filtered outputs may be summed to generate an antinoise signal that is in input to a loudspeaker within the headphone earpiece that substantial suppresses the ambient noise before it can be perceived by an end user of the headphones.

Abstract: The disclosed active noise suppression headphone system is directed to a headphone system that is capable of substantially suppressing high or low frequency interfering noise that penetrate through a headphone earpiece from multiple directions. An external microphone mounted with a housing of a headphone earpiece senses ambient noise outside of the earpiece. The sensed ambient noise may be processed through at least one parallel filter bank arranged in at least one headphone earpiece. Each parallel filter bank may include adaptively linked filters. The output of these filters may be amplified based on weighting factors that are dependent upon the sensed ambient noise and that are generated by a filtered x least mean square circuit. The amplified filtered outputs may be summed to generate an antinoise signal that is in input to a loudspeaker within the headphone earpiece that substantial suppresses the ambient noise before it can be perceived by an end user of the headphones.

Abstract: A method conceals dropouts in one or more audio channels of a multi-channel arrangement. The method maps transmitted signals into a frequency domain during an error-free signal transmission of two or more channels. A magnitude spectra and spectral filter coefficients are derived. The spectral filter coefficients relate the magnitude spectrum of the audio channel to the magnitude spectrum of at least one other channel. When a dropout occurs, a replacement signal is generated through the filter coefficients and a substitution signal. The filter coefficients may be generated prior to the detection of the dropout.

Abstract: A method conceals dropouts in one or more audio channels of a multi-channel arrangement. The method maps transmitted signals into a frequency domain during an error-free signal transmission of two or more channels. A magnitude spectra and spectral filter coefficients are derived. The spectral filter coefficients relate the magnitude spectrum of the audio channel to the magnitude spectrum of at least one other channel. When a dropout occurs, a replacement signal is generated through the filter coefficients and a substitution signal. The filter coefficients may be generated prior to the detection of the dropout.

Abstract: Recorded audio data is compressed to obtain a condensed version, by first selecting a number of subsequent non-overlapping segments of the audio data, then reducing each segment by temporal compression and combining the reduced segments into a shortened version which can be output. The temporal compression may be made with a local compression factor which varies between the segments. The segmenting may be chosen based on an innovation signal derived from the audio data itself to indicate a content change rate in the audio data.