摘要:
The invention is related to lossless encoding of a source signal, using a lossy encoded data stream and a lossless extension data stream which together form a lossless encoded data stream 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 de-correlation and time domain de-correlation 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 to a lossy decoder. The lossy encoder provides a lossy bit stream. The difference signal between the PCM signal and the lossy decoder output is lossless encoded, 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.
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 decorrelation 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. The at least two data streams resulting from the combination of lossy coding format with a lossless coding extension are the base layer containing the lossy coding information and the enhancement data stream for rebuilding the mathematically lossless original input signal. Furthermore 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:
Lossless audio coding performs decorrelation and encodes the transformed signal. The encoded bit stream comprises de-correlation parameters and the lossless representation data of the transformed signal. However, in the case of lossy based lossless coding, the additional amount of information exceeds the base layer amount of data. Therefore the additional data cannot be packed completely into the base layer e.g. as ancillary data. The data streams resulting from the combination of lossy coding format with a lossless coding extension are the base layer containing the lossy coding information and the enhancement data stream for rebuilding the mathematically lossless original input signal. 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. Assignment information data identify the different layers.
摘要:
In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The lossy decoder also provides side information that is used to control the coefficients of a prediction filter that de-correlates the difference signal between the PCM signal and the lossy decoder output. The de-correlated difference signal is lossless encoded, providing an extension bit stream. 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:
In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The difference signal between the PCM signal and the lossy decoder output is lossless encoded, 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.
摘要:
In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The lossy decoder also provides side information that is used to control the coefficients of a prediction filter that de-correlates the difference signal between the PCM signal and the lossy decoder output. The de-correlated difference signal is lossless encoded, providing an extension bit stream. 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:
For the quantisation of spectral data in an audio transform encoder psycho-acoustic information is required, i.e. an approximation of the true masking threshold. According to the invention, for each spectrum to be quantised in the audio signal encoding, an excitation pattern is computed and coded for both long and short window/transform lengths. The excitation patterns are grouped together in a variable-size matrix. A pre-determined sorting order with a fixed number of values only is applied to the excitation pattern data matrix values, and by that re-ordering a quadratic matrix is formed to which matrix' bit planes a SPECK encoding is applied.
摘要:
A method and apparatus for approximating a true masking threshold for the quantization of spectral data in an audio transform encoder. According to the invention, for each spectrum to be quantised in the audio signal encoding, an excitation pattern is computed and coded for both long and short window/transform lengths. The excitation patterns are grouped together in a variable-size matrix. A pre-determined sorting order with a fixed number of values only is applied to the excitation pattern data matrix values, and by that re-ordering a quadratic matrix is formed to which matrix' bit planes a SPECK encoding is applied.
摘要:
At the time of encoding audio content, the finally required data rate for delivery to the customer may be unknown. A data format is disclosed that is optimized for serving as Intermediate Format for efficient and fast recoding, to obtain one or more standard complying lossy encoded data streams with flexible data rates. Encoding can be performed in two steps that are inter-coordinated for cooperating, but may be locally and/or temporally separate. Between the partial encoders encoding parameters and/or auxiliary data are transmitted in a separate parameter enhancement layer, which complements a lossy data stream and can be used by the second encoder or transcoder for fast and computationally efficient implementation of the second encoding step. An additional lossless enhancement layer allows lossless reconstruction.