Abstract: Mixing N information-time signals. The time signals are each converted into the frequency domain, into one of N complex information signals where N is an integer greater than 1. Spectral values of the N complex information signals which match in a frequency are each converted into a first and a second component. The N first components of the N frequency-matching spectral values are combined into a first combination component. The N second components of the N frequency-matching spectral values are combined into a second combination component. The first and second combination components are combined into a result spectral value. The above steps are also performed for other frequency-matching spectral values of the N complex information signals for generating other result spectral values. The result spectral values thus obtained are combined into a complex output information signal.

Abstract: The invention relates to a method and an apparatus for mixing N information-time signals (s1(t), s2(t), . . . ) which are first each converted into the frequency domain, into one of N complex information signals (v1(f, t1), v2(f, t1), . . . ), where N is an integer greater than 1. The following steps are carried out. Spectral values of the N complex information signals which match in a frequency are each converted into a first and a second component (208). The N first components of the N frequency-matching spectral values are combined into a first combination component (210). The N second components of the N frequency-matching spectral values are combined into a second combination component (212). The first and second combination components are combined into a result spectral value (214). The above steps are also performed for other frequency-matching spectral values of the N complex information signals for generating other result spectral values (216, 220).

Abstract: A dynamic range compressor includes an input terminal for receiving input signal to be compressed, an amplifier unit for amplifying the signal to be compressed by an amplification factor, for deriving a compressed output signal, an output terminal for supplying the compressed output signal, a first envelope detector unit for deriving a first envelope signal from the input signal, and an amplifier control unit for generating an amplifier control signal in dependence of an envelope signal. Further including a second envelope detector unit for deriving a second envelope signal from the input signal, a first signal level prediction unit for generating a first prediction signal from the first envelope signal, a second signal level prediction unit for generating a second prediction signal from the second envelope signal, and a signal combination unit for combining the first and second prediction signals to generate a combined prediction signal.

Abstract: In order to realize a correction for changes in the reproduction loudness at low frequencies in a downmix-arrangement, a mixing arrangement is proposed for mixing at least two audio signals, which mixing arrangement is provided with a first unit (104) for deriving a first power signal, which is a measure for the power of the first audio signal, a second unit (105) for deriving a second power signal, which is a measure for the power of the second audio signal, a cross-correlation unit (103) for deriving a cross-correlation signal, which is a measure for a cross-correlation between the first and the second audio signal, a unit (106) for deriving multiplication parameters from the first and second power signals and the cross-correlation signal, and a multiplication and combination unit (107) for carrying out a signal processing on the first and second audio signals and combining them.

Abstract: A method for down-mixing of a m-channel audio signal (L, R, C, Ls, Rs, Rss, Lss) into a n-channel audio signal (Ro, Lo, Rso, Lso), where m is an integer for which holds m>n and n is an integer for which holds n?2, including the step of generating one of the n-channel audio signals of one side (right or left) of a listener (Ro, Lo, Rso, Lso), by a combination of: a first term including a signal component (R, L, Rs, Ls) of the m-channel audio signal of the same side only, and a second term dependent of m, including one or more of further signal components of the m-channel audio signal (C, Ls, Rs, Rss, Lss) of the same side only, multiplied by at least one respective filtering function (H1, H2, H3, H4, H5, H6, H7, H8), said filtering function being dependent on: a frequency characteristic of the transmission path between the position of the loudspeaker of the respective signal component of the further m-channel audio signal, and a position of the right ear or left ear, respectively, of a listener in an m-chan

Abstract: A microphone arrangement with improved directional characteristics is provided with at least two microphones (100, 102) and a signal processing arrangement (105). The signal processing arrangement is provided with a first (108) and a second input (109) for receiving the microphone signals of the at least two microphones. The inputs (108,109) are coupled to signal inputs of a first (110) and a second (111) multiplication circuit. The multiplication circuits are provided with control inputs for receiving respective first and second control signals, and with signal outputs. A control signal generator (112) is provided for generating the first and second control signals for the multiplication circuits (110,111). An arrangement (114) for a power corrected summation is provided, having a first and a second input coupled to the outputs of the first and second multiplication circuit, respectively, and having an output.

Abstract: A dynamic range compressor of a subband type for carrying out a dynamic compression on a broadband input signal includes a subband splitting device for splitting the broadband input signal into K narrowband subband signals. An amplifier unit amplifies each of the K subband signals to obtain K amplified subband signals. Further, a subband combining device is provided for combining the K amplified subband signals to obtain a broadband output signal, which is a dynamically compressed version of the broadband input signal. An envelope detecting device generates, for each of the K subbands, a respective one of K envelope signals. An amplifier control device generates, in dependence of the K envelope signals, K amplifier control signals, each being representative of one of the K amplification factors. The amplifier control device is adapted to generate an amplification control signal in dependence of more than one of the K envelope signals.

Abstract: A dynamic range compressor includes an input terminal for receiving input signal to be compressed, an amplifier unit for amplifying the signal to be compressed by an amplification factor, for deriving a compressed output signal, an output terminal for supplying the compressed output signal, a first envelope detector unit for deriving a first envelope signal from the input signal, and an amplifier control unit for generating an amplifier control signal in dependence of an envelope signal. Further including a second envelope detector unit for deriving a second envelope signal from the input signal, a first signal level prediction unit for generating a first prediction signal from the first envelope signal, a second signal level prediction unit for generating a second prediction signal from the second envelope signal, and a signal combination unit for combining the first and second prediction signals to generate a combined prediction signal.

Abstract: An interpolation circuit for interpolating a first and a second microphone signal and for generating an interpolated microphone signal includes a first input (100) for receiving the first microphone signal (am), a second input (101) for receiving the second microphone signal (am+1), an output (102) for outputting the interpolated microphone signal (s), a control input (103) for receiving a control signal (r), and a first circuit branch (104) including first (105) and second (106) inputs coupled to the first (100) and the second (101) input, respectively, of the interpolation circuit, and an output (107) coupled to the output (102) of the interpolation circuit, wherein the first circuit branch is provided with a means (108) for power-specific summing of the signals supplied to the first and second inputs of the first circuit branch and for outputting a power-specific summation signal at the output (107) of the first circuit branch (104).

Abstract: A sub-band splitter unit for splitting a broadband input signal in K narrowband sub-band signals, wherein K is an integer larger than 1, which sub-band splitter unit is provided with an input terminal for receiving the broadband input signal and K?1 sub-band filter circuits, each of the sub-band filter circuits is provided with an input and a first and a second output, a first filter arrangement coupled between the input and the first output, a second filter arrangement coupled between the input and the second output, and the sub-band splitter unit is provided with K output terminals for supplying the K sub-band signals. The first output of a k-th sub-band filter circuit is coupled to an input (103.k+1) of the (k+1)-th sub-band filter circuit. The input of the first sub-band filter circuit is coupled to the input of the sub-band splitter unit.

Abstract: A microphone arrangement with improved directional characteristics is provided with at least two microphones (100, 102) and a signal processing arrangement (105). The signal processing arrangement is provided with a first (108) and a second input (109) for receiving the microphone signals of the at least two microphones. The inputs (108,109) are coupled to signal inputs of a first (110) and a second (111) multiplication circuit. The multiplication circuits are provided with control inputs for receiving respective first and second control signals, and with signal outputs. A control signal generator (112) is provided for generating the first and second control signals for the multiplication circuits (110,111). An arrangement (114) for a power corrected summation is provided, having a first and a second input coupled to the outputs of the first and second multiplication circuit, respectively, and having an output.

Abstract: A method for down-mixing of a m-channel audio signal (L, R, C, Ls, Rs, Rss, Lss) into a n-channel audio signal (Ro, Lo, Rso, Lso), where m is an integer for which holds m>n and n is an integer for which holds n?2, including the step of generating one of the n-channel audio signals of one side (right or left) of a listener (Ro, Lo, Rso, Lso), by a combination of: a first term including a signal component (R, L, Rs, Ls) of the m-channel audio signal of the same side only, and a second term dependent of m, including one or more of further signal components of the m-channel audio signal (C, Ls, Rs, Rss, Lss) of the same side only, multiplied by at least one respective filtering function (H1, H2, H3, H4, H5, H6, H7, H8), said filtering function being dependent on: a frequency characteristic of the transmission path between the position of the loudspeaker of the respective signal component of the further m-channel audio signal, and a position of the right ear or left ear, respectively, of a listener in an m-chan

Abstract: A dynamic range compressor of the subband type for carrying out a dynamic compression on a broadband input signal includes a subband splitting device (102) for splitting the broadband input signal into K narrowband subband signals (SSB1, . . . SSBk, . . . , SSBK), where K is an integer larger than 1. An amplifier unit (104) is provided for amplifying each of the K subband signals by a respective amplification factor (A1, . . . AK) to obtain K amplified subband signals. Further, a subband combining device (105) is provided for combining the K amplified subband signals to obtain a broadband output signal, which is a dynamically compressed version of the broadband input signal. An envelope detecting device (107) is provided for generating, for each of the K subbands, a respective one of K envelope signals.

Abstract: A sub-band splitter unit for splitting a broadband input signal (G1), e.g., an audio signal, in K narrowband sub-band signals (L1, . . . , Lk, . . . . , LK), wherein K is an integer larger than 1, which sub-band splitter unit (100) is provided with an input terminal (100) for receiving the broadband input signal and K?1 sub-band filter circuits (SBF1, . . . , SBFk, . . . , SBFK?1), each of the sub-band filter circuits (SBFk) is provided with an input (103.k) and a first (104.k) and a second output (105.k), a first filter arrangement (LPFk) coupled between the input (103.k) and the first output (104.k), a second filter arrangement (HPFk) coupled between the input (103.k) and the second output (105.k), and the sub-band splitter unit is provided with K output terminals (102.1, . . . , 102.k, . . . , 102.K) for supplying the K sub-band signals. The first output (104.k) of a k-th sub-band filter circuit (SBFk) is coupled to an input (103.k+1) of the (k+1)-th sub-band filter circuit. The input (103.

Abstract: In order to compensate tonal changes arising from a multi-path propagation of sound portions during the mixing of multi microphone audio recordings as far as possible it is suggested to form spectral values of respectively overlapping time frames of samples of each a first microphone signal (100) and a second microphone signal (101). The spectral values (300) of the first microphone signal (100) are distributed with formation of spectral values (311) of a first sum signal to the spectral values (301) of a second microphone signal (101) in a first summing level (310), whereat a dynamic correction of the spectral values (300, 301) of one of the two microphone signals (100, 101) occurs. Spectral values (399) of a result signal are formed out of the spectral values (311) of the first sum signal which are subject to an inverse Fourier-transformation and a block junction (FIG. 3).

Abstract: In a peer-to-peer (P2P) system, in each individual transmitting peer (10), the incoming data stream is divided logically and with regard to time into different parts, which are buffered in a volatile memory (13) and are transmitted to the receiving peers (20, 30) on different paths by means of a peer-to-peer transmission mechanism (40). In a receiving peer (20), the received parts of the subdivided data stream are buffered in a volatile memory (21) and are reassembled into a complete data stream. Further, in the receiving peer (20), the parts of the subdivided data stream, buffered by the volatile memory (21), are copied into a persistent memory (22), from where they are copied back into the volatile memory (21) at a later point in time if necessary. A fraction of the received data is excluded from persistent storing by selecting parts of the subdivided data stream.

Abstract: In order to realize a correction for changes in the reproduction loudness at low frequencies in a downmix-arrangement, a mixing arrangement is proposed for mixing at least two audio signals, which mixing arrangement is provided with a first unit (104) for deriving a first power signal, which is a measure for the power of the first audio signal, a second unit (105) for deriving a second power signal, which is a measure for the power of the second audio signal, a cross-correlation unit (103) for deriving a cross-correlation signal, which is a measure for a cross-correlation between the first and the second audio signal, a unit (106) for deriving multiplication parameters from the first and second power signals and the cross-correlation signal, and a multiplication and combination unit (107) for carrying out a signal processing on the first and second audio signals and combining them.

Abstract: A method of generating an audio output signal according to a downward compatible sound format, the method including: generating a sum signal by combining a first input channel signal with a second input channel signal; and dynamically correcting the sum signal using samples of the first and second input channel signals from overlapping time windows.

Abstract: An interpolation circuit for interpolating a first and a second microphone signal and for generating an interpolated microphone signal includes a first input (100) for receiving the first microphone signal (am), a second input (101) for receiving the second microphone signal (am+1), an output (102) for outputting the interpolated microphone signal (s), a control input (103) for receiving a control signal (r), and a first circuit branch (104) including first (105) and second (106) inputs coupled to the first (100) and the second (101) input, respectively, of the interpolation circuit, and an output (107) coupled to the output (102) of the interpolation circuit, wherein the first circuit branch is provided with a means (108) for power-specific summing of the signals supplied to the first and second inputs of the first circuit branch and for outputting a power-specific summation signal at the output (107) of the first circuit branch (104).

Abstract: In order to reduce the disturbing background noises that may arise during the summation with weighting of the spectral coefficients using a correction factor in a downmix method, the proposition is made that the correction factors m(k) are computed as follows: eA(k)=Real(A(k))Real(A(k))+Imag(A(k))·Imag(A(k)) eB(k)=Real(B(k))·Real(B(k))+Imag(B(k))·Imag(B(k)) x(k)=Real(A(k))·Real(B(k))+Imag(A(k))·Imag(B(k)) w(k)=D·x(k)/(eA(k)+L·eB(k)) m(k)=(w(k)2+1)(1/2)?w(k) wherein m is the kth correction factor; and A(k) is the kth spectral value of the signal to be prioritized; and B(k) is the kth spectral value of the signal not to be prioritized; and D is the degree of compensation; and L is the degree of the limitation of the compensation.