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
  • G10L19/002—Dynamic bit allocation
• • 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
  • G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/0212—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 spectral analysis, e.g. transform vocoders or subband vocoders using orthogonal transformation
• • 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
  • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
  • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders

Definitions

• This vector hd is subtracted from the vector tp by the subtracting unit 90.
• the result is tt, the target vector for transform coding.
• the resulting normalized FFT coefficient vector is partitioned into 3 frequency bands: (1) the low-frequency band consisting of the first 6 normalized FFT coefficients (i.e. from 0 to 1250 Hz), (2) the mid-frequency band consisting of the next 10 normalized FFT coefficients (from 1500 to 3750 Hz), and (3) the high-frequency band consisting of the remaining 17 normalized FFT coefficients (from 4000 to 8000 Hz).
• processor 130 uses a "greedy” algorithm to perform adaptive bit allocation.
• the technique is “greedy” in the sense that it allocates one bit at a time to the most "needy" frequency component without regard to its potential influence on future bit allocation.
• the LPC power spectrum is assumed to be the power spectrum of the coding noise.
• the noise loudness at each of the 33 frequencies of a 64-point FFT is estimated using the masking threshold calculated above and a simplified version of the noise loudness calculation method in Schroeder et al.
• the simplified noise loudness at each of the 33 frequencies is calculated as follows. First, the critical bandwidth Bj at the i-th frequency is calculated using linear inte ⁇ olation of the critical bandwidth listed in table 1 of Scharf s book chapter in Tobias. The result is the approximated value of the term df/dx in equation (3) of Schroeder et al.
• the 33 critical bandwidth values are pre-computed and stored in a table. Then, for the i-th frequency, the noise power N f is compared with the masking threshold Mj. If N, ⁇ M
• the transform coefficient quantizer 120 quantizes the transform coefficients contained in tc using the bit allocation signal ba.
• the DC term of the FFT is a real number, and it is scalar quantized if it ever receives any bit during bit allocation.
• the maximum number of bits it can receive is 4.
• a conventional two-dimensional vector quantizer is used to quantize the real and imaginary parts jointly.
• the maximum number of bits for this 2-dimension VQ is 6 bits.
• a conventional 4-dimensional vector quantizer is used to jointly quantize the real and imaginary parts of two adjacent FFT coefficients.
• the resulting VQ codebook index array IC contains the main information of the TPC encoder. This index array IC is provided to the multiplexer 180, where it is combined with side information bits. The result is the final bit-stream, which is transmitted through a communication channel to the TPC decoder.
• the transform coefficient quantizer 120 also decodes the quantized values of the normalized transform coefficients. It then restores the original o gain levels of these transform coefficients by multiplying each of these coefficients by the corresponding elements of mag and the quantized linear gain of the corresponding frequency band. The result is the output vector dtc.
• FIG. 2 An illustrative decoder embodiment of the present invention is shown 15 in Figure 2.
• the demultiplexer 200 separates all main and side information components from the received bit-stream.
• the main information the transform coefficient index array IC, is provided to the transform coefficient decoder 235.
• adaptive bit allocation must be performed to determine how many of the main 20 information bits are associated with each quantized transform coefficient.
• the transform coefficient decoder 235 can then correctly decode the main information and obtain the quantized versions of the normalized transform coefficients.
• the decoder 235 also decodes the gains using the gain index array IG. For each subframe, there are two gain indices (5 and 7 bits), which are decoded into the quantized log gain of the low-frequency band and the quantized versions of the level-adjusted log gains of the mid-and high-frequency log gains. The quantized low-frequency log gain is then added back to the quantized versions of the level-adjusted mid- and high-frequency log gains to obtain the quantized log gains of the mid- and high-frequency bands.
• the high-frequency synthesis processor 240, inverse transform processor 245, and the inverse shaping filter 250 are again exact replicas of the corresponding blocks (140, 150, and 160) in the TPC encoder. Together they perform high-frequency synthesis, noise fill-in, inverse transformation, and inverse shaping filtering to produce the quantized excitation vector et.
• the adder 255 adds dh and et to get dt, the quantized version of the LPC prediction residual d.
• This dt vector is fed back to the pitch predictor inside block 210 to update its internal storage buffer for dt (the filter memory of the pitch predictor).
• the long-term postfilter 260 is basically similar to the long-term postfilter used in the ITU-T G.728 standard 16 kb/s Low-Delay CELP coder.
• the main difference is that it uses ⁇ b lk , the sum of the three quantized pitch i-l taps, as the voicing indicator, and that the scaling factor for the long-term postfilter coefficient is 0.4 rather than 0.15 as in G.728. If this voicing indicator is less than 0.5, the postfiltering operation is skipped, and the output vector fdt is identical to the input vector dt. If this indicator is 0.5 or more, the postfiltering operation is carried out.
• the LPC synthesis filter 265 is the standard LPC filter — an all-pole, direct-form filter with the quantized LPC coefficient array a. It filters the signal fdt and produces the long-term postfiltered, quantized speech vector st.
• This st vector is passed through the short-term postfilter 270 to produce the final TPC decoder output speech signal fst.
• this short-term postfilter 270 is very similar to the short-term postfilter used in G.728. The only differences are the following. First, the pole-controlling factor, the zero-controlling factor, and the spectral-tilt controlling factor are 0.7, 0.55, and 0.4, respectively, rather than the corresponding values of 0.75, 0.65, and 0.15 in G.728. Second, the coefficient of the first-order spectral-tilt compensation filter is linearly interpolated sample-by-sample between frames. This helps to avoid occasionally audible clicks due to discontinuity at frame boundaries.

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

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Citations (2)

• 1997
  • 1997-02-26 JP JP9530382A patent/JPH11504733A/ja active Pending
  • 1997-02-26 WO PCT/US1997/002898 patent/WO1997031367A1/en not_active Application Discontinuation
  • 1997-02-26 CA CA 2219358 patent/CA2219358A1/en not_active Abandoned
  • 1997-02-26 EP EP97907830A patent/EP0954851A1/en not_active Withdrawn
  • 1997-02-26 MX MX9708203A patent/MX9708203A/es unknown

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