Abstract: An apparatus for providing a set of spatial cues associated with an upmix audio signal having more than two channels on the basis of a two-channel microphone signal has a signal analyzer and a spatial side information generator. The signal analyzer is configured to obtain a component energy information and a direction information on the basis of the two-channel microphone signal, such that the component energy information describes estimates of energies of a direct sound component of the two-channel microphone signal and of a diffuse sound component of the two-channel microphone signal, and such that the directional information describes an estimate of a direction from which the direct sound component of the two-channel microphone signal originates. The spatial side information generator is configured to map the component energy information and the direction information onto a spatial cue information describing the set of spatial cues associated with an upmix audio signal having more than two channels.

Abstract: A digital audio broadcasting (DAB) communication system with a decoder buffer specified by a maximum number of encoded frames is disclosed. A predicted number of encoded frames, Fpred, in the decoder is sent to a receiver with audio data. If the decoder buffer level becomes too high, additional bits are allocated to each frame for each of N programs. If the decoder buffer level becomes too low, fewer bits are allocated to each frame for each of the N programs. Fpred can also be employed to (i) enable the decoder; and (ii) synchronize the transmitter and the receiver. The receiver fills the decoder buffer with Fpred frames before commencing decoding frames. The transmitter and receiver clocks may be synchronized with a feedback loop that compares the actual level of the decoder buffer to the predicted value, Fpred, received from the transmitter.

Abstract: The invention relates to an audio signal synthesizer, the audio signal synthesizer comprises a transformer for transforming the down-mix audio signal into frequency domain to obtain a transformed audio signal; a signal generator for generating a first auxiliary signal, for generating a second auxiliary signal, and for generating a third auxiliary signal upon the basis of the transformed audio signal; a de-correlator for generating a first de-correlated signal, and for generating a second de-correlated signal from the third auxiliary signal, the first de-correlated signal and the second de-correlated signal being at least partly de-correlated; and a combiner for combining the first auxiliary signal with the first de-correlated signal to obtain a first audio signal, and for combining the second auxiliary signal with the second de-correlated signal to obtain the second audio signal, the first audio signal and the second audio signal forming the multi-channel audio signal.

Abstract: A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. An audio encoder marks a frame as “dropped” whenever a buffer overflow might occur. Only a small number of bits are utilized to process a lost frame, thereby preventing the buffer from overflowing and allowing the encoder buffer-level to quickly recover from the potential overflow condition. The audio encoder optionally sets a flag that provides an indication to the receivers that a frame has been lost. If a “frame lost” condition is detected by a receiver, the receiver can optionally employ mitigation techniques to reduce the impact of the lost frame(s).

Abstract: An apparatus and a method for generating an acoustic signal with an enhanced spatial effect, said apparatus comprising a signal filter bank adapted to filter a difference audio signal with a filter characteristic to limit a bandwidth of said difference audio signal, wherein said bandwidth limited difference audio signal is applied to at least one pair of loudspeakers for dipole sound emission.

Abstract: A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The decoder buffer level limits are specified in terms of a maximum number of encoded frames (or duration). The transmitter can predict the number of encoded frames, Fpred, in the decoder buffer and transmit the value, Fpred, to the receiver with the audio data. If the transmitter determines that the decoder buffer level is becoming too high, the frames being generated by the encoder are too small and additional bits are allocated to each frame for each of the N programs. Likewise, if the transmitter determines that the decoder buffer level is becoming too low, the frames being generated by the encoder are too big and fewer bits are allocated to each frame for each of the N programs. The transmitted predicted buffer level, Fpred, can also be employed to (i) determine when the decoder should commence decoding frames; and (ii) synchronize the transmitter and the receiver.

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 and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The transmitter predicts the number of encoded frames, Fpred, in the buffer having a limited level and transmits the value, Fpred, to the receiver with the frame. If the transmitter determines that the decoder buffer level is high, the frames being generated by the encoder are small and additional bits are allocated to each frame for each of the N programs. Likewise, if the transmitter determines that the decoder buffer level is becoming low, the frames being generated by the encoder are big and fewer bits are allocated to each frame for each of the N programs. The transmitted predicted buffer level, Fpred, can also be employed to (i) determine when the decoder should commence decoding frames; and (ii) synchronize the transmitter and the receiver clock using feedback depending on the compared level of the decoder to the actual level to Fpred.

Abstract: A system and method for use in filtering of an acoustic signal are provided for producing an output signal of attenuated amount of diffuse sound in accordance with predetermined parameters of desired output directional response and required attenuation of diffuse sound. The system includes a filtration module and a filter generation module including a directional analysis module and filter construction module.

Abstract: Acoustic echo control and noise suppression in telecommunication systems. The proposed method of processing multi-channels audio loudspeakers signals and at least one microphone signal, comprises the steps of: transforming the input microphone signals (y1 (n), y2 (n), . . . , yM (n)) into input microphone short-time spectra, computing a combined loudspeaker signal short-time spectrum [X(i,k)] from the loudspeaker signals, (x1 (n), x2 (n), . . . , xL (n)), computing a combined microphone signal short-time spectrum [Y(i,k)] from the input microphone signal, (y1 (n), y2 (n), . . . , yM (n)), estimating a magnitude or power spectrum of the echo in the combined microphone signal short-time spectrum, computing a gain filter (G(i,k)) for magnitude modification of the input microphone short-time spectra, applying the gain filter to at least one of the input microphone spectra, converting the filtered input microphone spectra into the time domain (e1 (n), e2 (n), . . . , eM (n)).

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.

Abstract: An apparatus for generating an enhanced downmix signal on the basis of a multi-channel microphone signal has a spatial analyzer configured to compute a set of spatial cue parameters having a direction information describing a direction-of-arrival of a direct sound, a direct sound power information and a diffuse sound power information on the basis of the multi-channel microphone signal. The apparatus also has a filter calculator for calculating enhancement filter parameters in dependence on the direction information describing the direction-of-arrival of the direct sound, in dependence on the direct sound power information and in dependence on the diffuse sound power information. The apparatus also has a filter for filtering the microphone signal, or a signal derived therefrom, using the enhancement filter parameters, to obtain the enhanced downmix signal.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

Abstract: A preferred embodiment of an apparatus for computing filter coefficients for an adaptive filter for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal includes an extractor for extracting a stationary component signal or a non-stationary component signal from the loudspeaker signal or from a signal derived from the loudspeaker signal, and a computer for computing the filter coefficients for the adaptive filter on the basis of the extracted stationary component signal or the extracted non-stationary component signal.

Abstract: Methods and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. The disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. The distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value.

Abstract: A method of processing an audio signal is disclosed. The present invention comprises receiving downmix signal including object signals, transforming the downmix signal per frequency band, determining a direction of an object signal from the transformed downmix signal, and determining blind information by estimating a level of the object signal corresponding to the direction. Accordingly, the present invention generates blind information in case of using an encoder incapable of generating object information, thereby enabling a gain and panning of object to be controlled using the blind information.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: A binaural cue coding scheme involving one or more object-based cue codes, wherein an object-based cue code directly represents a characteristic of an auditory scene corresponding to the audio channels, where the characteristic is independent of number and positions of loudspeakers used to create the auditory scene. Examples of object-based cue codes include the angle of an auditory event, the width of the auditory event, the degree of envelopment of the auditory scene, and the directionality of the auditory scene.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: One or more attributes (e.g., pan, gain, etc.) associated with one or more objects (e.g., an instrument) of a stereo or multi-channel audio signal can be modified to provide remix capability. An audio decoding apparatus obtains an audio signal having a set of objects and side information. The apparatus obtains a set of mix parameters from a user input and an attenuation factor from the set of mix parameters. The apparatus then generates a plural-channel audio signal using at least one of the side information, the attenuation factor or the set of mix parameters.