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 digital compressor for compressing an input audio signal is presented. The digital compressor comprises a compression gain control for providing a compression gain parameter, and a compression parameter determiner for determining a compression ratio from the compression gain parameter. The compression parameter determiner may be configured to weight the compression gain parameter by a predetermined weight factor to obtain the compression ratio. The digital compressor further comprises an auxiliary signal generator for manipulating the input audio signal in dependence of the compression ratio to obtain a first auxiliary signal, and a combiner unit for combining the first auxiliary signal with the compression gain parameter to obtain a second auxiliary signal, and for combining the input audio signal with the second auxiliary signal to obtain the compressed audio signal.
Abstract:
According to the invention, a device for post-processing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal, an interchannel time difference between the channel signal and the downmix signal, and a classification indication indicating a transient type of the downmix signal; and a post-processor for post-processing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication and the interchannel time difference.
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:
A stereo decoding method and apparatus are disclosed. The method includes: restoring a monophonic signal from a received code stream through decoding; restoring an interchannel level difference, a group delay, and a group phase from the received code stream through decoding; and processing the monophonic signal according to the interchannel level difference, group delay, and group phase to obtain a first channel signal and a second channel signal. According to the stereo decoding method and apparatus provided in embodiments of the present invention, the first and second channel signals are obtained according to the monophonic signal, ILD, group delay, and group phase by referring to not only the ILD but also the group delay and group phase, thereby yielding favorable stereo sound field effect for the obtained first and second channel signals.
Abstract:
An audio compression system for compressing an input audio signal, and the audio compression system comprises a digital filter for filtering the input audio signal, where the digital filter comprises a frequency transfer function having a magnitude over frequency, where the magnitude is formed by an equal loudness curve of a human ear to obtain a filtered audio signal, and a compressor which is configured to compress the input audio signal upon the basis of the filtered audio signal to obtain a compressed audio signal.
Abstract:
The invention relates to a portable electronic device, comprising: a housing comprising at least one hole; and at least two directional microphones mounted in the housing and placed coincidentally for stereo sound pickup, each one of the microphones defining a main sound axis and each one of the two directional microphones defining a direct sound direction and a opposite sound direction which describe opposite directions of the main sound axis, wherein the at least one hole is a common hole shared between the at least two directional microphones such that the main sound axis of the at least two directional microphones are pointing through the common hole in different directions.
Abstract:
The invention relates to a method for processing a multi-channel audio signal which carries a plurality of audio channel signals. The method comprises determining a time-scaling position using the plurality of audio channel signals and time-scaling each audio channel signal of the plurality of audio channel signals according to the time-scaling position to obtain a plurality of time scaled audio channel signals.
Abstract:
The disclosure relates to an audio signal processing apparatus for processing an input audio signal, comprising a filter unit comprising a plurality of filters, each filter configured to filter the input audio signal to obtain a plurality of filtered audio signals, each filter designed according to an extended mode matching beamforming applied to a surface of a half revolution, the surface partially characterizing a loudspeaker enclosure shape, a plurality of scaling units, each scaling unit configured to scale the plurality of filtered audio signals using a plurality of gain coefficients to obtain a plurality of scaled filtered audio signals, and a plurality of adders, each adder configured to combine the plurality of scaled filtered audio signals, thereby providing an output audio signal for producing a sound field having a beam directivity pattern defined by the plurality of gain coefficients.
Abstract:
A sound processing node for an arrangement of sound processing nodes is disclosed. The sound processing nodes being configured to receive a plurality of sound signals, wherein the sound processing node comprises a processor configured to determine a beamforming signal on the basis of the plurality of sound signals weighted by a plurality of weights, wherein the processor is configured to determine the plurality of weights using a transformed version of a linearly constrained minimum variance approach, the transformed version of the linearly constrained minimum variance approach being obtained by applying a convex relaxation to the linearly constrained minimum variance approach.