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. In order to reduce the amount of data to be stored persistently, it is proposed that a fraction of the received data is excluded from persistent storing, wherein this exclusion is performed by selecting parts of the subdivided data stream.

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: Source coding digitized audio signals includes providing time or spectral domain sampling values of the digitized audio signal, requantizing the sampling values according their permissible quantizing noise as determined by a coding and requantizing control signal, and multiplexing the control signal and the requantized sampling values into a time multiplex frame depending on the bit rate reduction employed. The coding and requantizing control signal is determined from the sampling values by determining the global masking effect from all relevant maskers which are tonal maskers and noise maskers, and which result from the sampling values, and from a silence audio threshold.