Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
  • G10L19/16—Vocoder architecture
  • G10L19/18—Vocoders using multiple modes
  • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
• • H—ELECTRICITY
  • H03—ELECTRONIC CIRCUITRY
  • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
  • H03M7/00—Conversion of a code where information is represented by a given sequence or number of digits to a code where the same, similar or subset of information is represented by a different sequence or number of digits
  • H03M7/30—Compression; Expansion; Suppression of unnecessary data, e.g. redundancy reduction
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/0017—Lossless audio signal coding; Perfect reconstruction of coded audio signal by transmission of coding error
• • H—ELECTRICITY
  • H03—ELECTRONIC CIRCUITRY
  • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
  • H03M13/00—Coding, decoding or code conversion, for error detection or error correction; Coding theory basic assumptions; Coding bounds; Error probability evaluation methods; Channel models; Simulation or testing of codes
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing

Definitions

• FIG. 1 general block diagram of a lossy based lossless encoder
• FIG. 6 block diagram for a second embodiment encoder
• FIG. 7 block diagram for the first embodiment encoder with additional rounding gain factor processing
• Fig. 11 optional time de-correlation block or step for the encoder (left) and the decoder (right) ;
• the output signal of the first quantiser 12 passes through an interpolation and inverse sub-band filter bank block or step 18 and is subtracted in a second subtractor 152 from the correspondingly delayed (delay 17) input signal SpcM-
• the output signal of the second subtractor passes through a lossless coding TD (time domain) block or step 19 to multiplexer 10, which outputs a correspondingly multiplexed encoded bit stream S Q ⁇ Q .
• Blocks/stages 11, 12, 13 and 14 together form a lossy en- coder.
• a transform following the analysis sub-band filter bank 11 inside the lossy encoder is replaced by a rounding/quantisation step 12 and an integer transform 13.
• the lossy encoded bit stream and the encoded (integer) spectral and temporal residua may be multiplexed to form a single bit stream or to form two streams (lossy coded stream and lossless extension carrying the residua) or to form three streams, the lossy coded stream, the coded spectral residuum stream and the coded temporal residuum stream.
• the encoded bit stream S Q ⁇ Q enters a de-multi-plexer 20, which outputs a lossy encoded bit stream, a lossless encoded FD bit stream and a lossless en- coded TD bit stream.
• the lossy encoded bit stream passes through a decoding block or step 24 (that regains the quantised values from the index values representing the quantised values, i.e. the second output signal of quantiser 14), a first adder 25, an inverse integer transform block or step 23 and an interpolator and inverse sub-band filter bank block or step 21 to a first input of a second adder 28.
• the lossless encoded FD bit stream passes through a lossless decoder FD block or step 26 to the second input of the first adder.
• the lossless encoded TD bit stream passes through a lossless decoder TD block or step 29 and a corresponding delay 27 to the second input of the second adder, which outputs the decoded lossy PCM signal SpcM-
• the preferred embodiments will use the well-known mp3 standard as the embedded lossy core codec, the encoding part of which is shown in Fig. 3.
• the original input signal Sp Q jv [ passes through a polyphase filter bank & decimator 503, a segmentation & MDCT 504 and a bit allocation and quantiser 505 to multiplexer 507.
• Input signal Sp Q M also passes through an FFT stage or step 501 to a psycho-acoustic analysis 502 which controls the segmentation (or windowing) in step/stage 504 and the quantisation 505.
• the bit allocation and quantiser 505 also provides side information 515 that passes through a side info encoder 506 to multiplexer 507 which outputs signal 517.
• the mp3 standard applies nonuniform (i.e.
• the first embodiment encoder in Fig. 4 includes the mp3 standard encoder of Fig. 3. However, the embedded mp3 encoder has been modified in the following manner: The sub-band signals 512 from polyphase filter bank & deci- mator 503 are quantised (performing a rounding) and the original MDCT transform in block or step 504 of the individual sub-band signals has been replaced by an integer MDCT (Int-MDCT) transform 504.
• the integer MDCT approximates the numerical behavior of the original non-integer MDCT trans- form to guarantee that the embedded mp3 bit stream, produced by the Fig. 4 encoder, can be decoded by any standard- conform mp3 decoder without quality degradation.
• the inventive encoder signal processing comprises lossless encoding schemes for two error signals: in frequency domain and in time domain.
• the quantised transform coefficients obtained from the mp3 bit stream 5114 are rounded in an inverse quantiser rounding block or step 521 to obtain integer values which are subtracted from the original inte- ger MDCT transform coefficients 513 in a first subtractor 522.
• the resulting integer error values are encoded loss- lessly in a lossless encoding FD (frequency domain) block or step 523 and are multiplexed (by MUX 507) into the bit stream.
• the decoder for the second embodiment is identical to the first embodiment decoder.
• the depicted blocks have essentially the same function or operation like the corresponding blocks in Fig. 4.
• the sub-band signals 206 are multiplied in a first multiplier 223 by a gain factor g.
• integer-valued sub-band samples 226 are obtained that can be processed by the integer MDCT 248.
• the rounded sub-band samples 226 are to be divided by g in a first divider 230.
• the resulting samples 231 are quantised versions of the original sub-band samples 206.
• the quantised sample values are multiples of 1/g.
• Optional extension 2 decoding of lossy version (mp3 decoder) Because the full bit stream of the proposed lossy based lossless coding scheme comprises an embedded standard- conform mp3 bit stream, conventional mp3 decoding can be applied. In the corresponding decoder signal flow depicted in Fig. 9, the parts of the bit stream that describe the time domain residual 240 and the frequency domain residual 243 are discarded. The discarding operation can take place upstream the decoder as well, e.g. in a transmission network.
• the blocks or steps 301, 306, 217 and 232 are identical to the corresponding blocks or steps in Fig. 5. However, an in- verse MDCT block or stage 3081 may be used instead of an inverse integer MDCT block or stage 308.
• this optional decoder will not render lossless PCM samples, it is not necessary (but possible) to use an inverse integer MDCT 308 instead of an inverse MDCT 3081 in front of the interpolation & poly-phase filter bank 232.
• the information on the frequency domain residual 258 may be encoded using a multi-layered bit stream structure.
• the bit plane arithmetic coding principle known from the MPEG SLS draft standard or a similar scheme may be applied.
• a fully scalable coding scheme can be realised with fine granularity of the bit rate (and quality) steps .
• Optional extension 5 de-correlation of the time domain residual
• Fig. 5 and Fig. 6 the possibility to apply de-correlation of the time domain residual signal at encoder side and corresponding inverse de-correlation of the time domain residual signal at decoder side has been mentioned above.
• Fig. 11 shows which functions may be inside block 527 (left side) and block 303 (right side) .
• a linear prediction filter is applied to the time domain residual.
• the prediction signal is subtracted from that residual to remove any remaining short term correlation.
• the pre- diction filter coefficients are adapted to the signal characteristics by analysing information from the lossy encoder and/or from the frequency domain de-correlation block of the codec core, and, optionally, the actual residual signal is analysed. Further, it may be necessary to signal information on the filter adaptation to the inverse time domain de- correlation block in the receiver or decoder. Therefore, it may be necessary to include certain information in the lossless extension of the gross bit stream.
• the invention achieves lossless coding based on existing lossy audio coding schemes with hybrid filter banks, like mp3.
• the only non-trivial signal processing block that should have a platform-independent implementation is the poly-phase synthesis filter bank.
• the first and second embodiments have specific advantages: the first embodiment allows for low-complexity implementa- tion of the encoder because only one set of (integer) MDCT transforms is to be computed in the encoder. On the other hand, the second embodiment allows for a higher-quality version of the encoder, where the embedded mp3 bit stream is produced by an unmodified mp3 encoder, at the cost of com- puting two sets of MDCT transforms in parallel.

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• Engineering & Computer Science (AREA)
• Physics & Mathematics (AREA)
• Human Computer Interaction (AREA)
• Signal Processing (AREA)
• Health & Medical Sciences (AREA)
• Audiology, Speech & Language Pathology (AREA)
• Computational Linguistics (AREA)
• Acoustics & Sound (AREA)
• Multimedia (AREA)
• Quality & Reliability (AREA)
• Theoretical Computer Science (AREA)
• Probability & Statistics with Applications (AREA)
• Compression, Expansion, Code Conversion, And Decoders (AREA)
• Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

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• 2006
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