摘要:
Lossless compression algorithms can only exploit redundancies of the original audio signal to reduce the data rate, but not irrelevancies as identified by psycho-acoustics. Lossless audio coding schemes apply a filter or transform for de-correlation and then encode the transformed signal. The encoded bit stream comprises the parameters of the transform or filter, and the lossless representation of the transformed signal. However, in case of lossy based lossless coding the additional amount of information exceeds the amount of data for the base layer by a multiple of the base layer data amount. Therefore the additional data cannot be packed completely into the base layer data stream e.g. as ancillary data. Several intermediate quality layers are possible. However, these data streams are not independent from each other. Every higher layer depends on the lower layers and can only be reasonably decoded in combination with these lower layers. According to the invention, a special combination of one-time header information with repeated header information in a block structure is used, which kind of combination depends on the type of application. Assignment information data identify the different parts or layers of the lossless format belonging to one input signal. Synchronisation data are used to combine the different data streams or parts or layers to a single lossless or intermediate output signal. These features are used in a file format and in a streaming format.
摘要:
Advanced solutions for encrypting multi-layer audio data are required, ie. audio data that comprise a base layer and one or more enhancement layers. A method for encrypting such an encoded audio signal comprises separating the base layer into two sections, encrypting the side information within frames of the second section of the base layer, and encrypting at least a part of the data of the enhancement layer, wherein the encrypted section of the base layer and the encrypted enhancement layer require different decryption keys for decryption. Thus, free preview zones are possible to implement.
摘要:
In lossy based lossless coding a PCM audio signal passes through a lossy encoder (101) to a lossy decoder (102). The lossy encoder provides a lossy bit stream (111). The lossy decoder also provides side information (115) that is used to control (105) the coefficients (118) of a prediction filter (106) that de-correlates the difference signal (104) between the PCM signal and the lossy decoder output. The de-correlated difference signal is lossless encoded (108), providing an extension bit stream (121). Instead of, or in addition to, de-correlating in the time domain, a de-correlation in the frequency domain using spectral whitening can be performed. The lossy encoded bit stream together with the lossless encoded extension bit stream form a lossless encoded bitstream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/decoding.
摘要:
The present invention provides method and device for transcoding between audio coding formats with different time-frequency analysis domains, as used for example by MPEG-AAC and mp3, particularly for facilitated and faster transcoding between such audio signals. A method for transcoding a framed audio signal from a first parameter domain (PD A ) into a second parameter domain (PD 3 ) comprises linearly transforming (T T ) two or more parameters of the first parameter domain (PD A ) to at least one parameter of the second parameter domain (PD B ), wherein the two or more parameters of the first parameter domain come from different frames of the audio signal in the first parameter domain. The linear transformation (T T ) can be described as a matrix and implemented as a look-up table.
摘要:
The invention is related to lossless encoding of a source signal (S PCM ), using a lossy encoded data stream and a lossless extension data stream which together form a lossless encoded data stream (S ENC ) for said source signal, whereby lossless audio compression means audio coding with bit-exact reproduction of the original PCM samples at decoder output. The lossy encoding/decoding may be an mp3 coding/decoding. The invention uses an integer MDCT and frequency domain decorre-lation (16) and time domain de-correlation (16) for the residual signal of the base-layer lossy audio codec. The exploitation of side information from the lossy base-layer codec allows for reduction of redundancies in the gross bit stream, thus improving the coding efficiency of the lossy based lossless codec.
摘要:
In lossy based lossless coding a PCM audio signal passes through a lossy encoder (41) to a lossy decoder (42). The lossy encoder provides a lossy bit stream. The difference signal (S Diff ) between the PCM signal and the lossy decoder output is lossless encoded (52), providing an extension bit stream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform using enhanced de-correlation, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/de-coding.