WO2000033293A1 - Fixed-point multiplication for adpcm speech coder - Google Patents
Fixed-point multiplication for adpcm speech coder Download PDFInfo
- Publication number
- WO2000033293A1 WO2000033293A1 PCT/SG1998/000098 SG9800098W WO0033293A1 WO 2000033293 A1 WO2000033293 A1 WO 2000033293A1 SG 9800098 W SG9800098 W SG 9800098W WO 0033293 A1 WO0033293 A1 WO 0033293A1
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- WO
- WIPO (PCT)
- Prior art keywords
- word length
- values
- fixed point
- predictor
- signal values
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Classifications
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
Definitions
- This invention relates to the implementation of a digital speech coder for the transmission of speech or voice band data over a communications network.
- ADPCM Adaptive Differential Pulse Coded Modulation
- the ADPCM algorithm achieves speech compression by combining adaptive quantization and differential PCM.
- Adaptive quantization adjusts the step size of the quantizer as the signal changes. This allows the algorithm to accommodate variations in the signal amplitude.
- Differential PCM involves transmitting the difference between the current and previous signal sample instead of simply transmitting the current sample itself. The difference signal obtained in this way tends to have a much lower dynamic range compared to the original signal and may therefore be quantized to a specific signal-to-noise ratio with fewer bits.
- the difference signal is computed from the current signal sample and a signal estimate determined by an adaptive predictor.
- the adaptive predictor uses signal estimates of previous samples to obtain an approximation of the current sample. This is performed in both the encoder and decoder so that they are synchronised with each other and there will not be any accumulation of errors in the reconstructed signal at the decoder output.
- the adaptive predictor is represented by a two-pole, six- zero adaptive predictive filter.
- the combination of poles and zeros enables the filter to deal with any general input signal.
- the sixth-order all-zero filter is needed to stabilise the filter and prevent it from drifting into oscillation.
- the filter coefficients are updated based on a simplified gradient algorithm.
- the signal estimate is computed by:
- the range of values of the predictor coefficients is limited to ⁇ 2 and are stored as 16-bit fixed point values.
- the quantized difference signal and reconstructed signal can vary between -32768 to 32767. Initially 16-bit fixed point values, they are then converted to floating point and stored.
- the aforementioned ITU-T recommendation specifies that the multiplication operation should be performed in floating point, by converting all inputs to floating point values with 6 bits of mantissa and 4 bits of exponent. The resulting product is then converted back into a 16-bit fixed point number.
- a method for encoding speech or voice band data by way of adaptive differential pulse coded modulation including an adaptive predictor procedure which implements an adaptive predictive filter for generating a signal estimate from quantized difference signal values, reconstructed signal values and respective predictor coefficients according to a predetermined multiplication and accumulation operation, wherein the quantized difference signal values and reconstructed signal values are represented by single word length fixed point binary values, including performing multiplication in fixed point format between the respective said predictor coefficients and the quantized difference signal values and reconstructed signal values to generate respective double word length fixed point partial product values, summing the double word length fixed point partial product values to form a double word length predictor sum and rounding the predictor sum to a single word length fixed point representation of said signal estimate.
- the present invention also provides An adaptive differential pulse coded modulation encoder for encoding speech or voice band data for transmission over a communications network, including an adaptive predictor having an adaptive predictive filter for generating a signal estimate from input quantized difference signal values, input reconstructed signal values and respective predetermined predictor coefficients, wherein the quantized difference signal values and reconstructed signal values are represented by single word length fixed point binary values, the adaptive predictive filter including a multiplier which performs multiplication in fixed point format between the respective said predictor coefficients and the quantized difference signal values and reconstructed signal values to generate respective double word length fixed point partial product values, and an accumulator for slimming the double word length fixed point partial product values to form a double word length predictor sum and rounding the predictor sum to a single word length fixed point representation of said signal estimate.
- an adaptive predictor having an adaptive predictive filter for generating a signal estimate from input quantized difference signal values, input reconstructed signal values and respective predetermined predictor coefficients, wherein the quantized difference signal values and reconstructed signal values are represented
- the single word length representations comprise 16 bit binary values and the double word length representations comprise 32 bit binary values.
- the double word length representations comprise 32 bit binary values.
- the present invention relates to adaptive differential pulse coded modulation (ADPCM) of speech or voice band data for transmission over a communications network, of the type which is described in ITU-T Recommendation G.726, the disclosure of which is incorporated herein by reference.
- ADPCM encoder in the ITU-T recommendation converts a 64 kbit/s PCM input into an ADPCM compressed output for transmission.
- the accompanying drawing figure illustrates a block diagram of an ADPCM encoder according to the ITU-T recommendation. Referring to the figure, an A-law or ⁇ -law PCM input stream is first converted to uniform PCM. A difference signal is then obtained by subtracting an estimate of the input signal from the input signal itself.
- An adaptive quantizer is used to assign a quantized value of a predetermined number of binary digits to the value of the difference signal for transmission to the decoder.
- An inverse adaptive quantizer is arranged to produce a quantized difference signal from the quantized value output from the adaptive quantizer.
- the input signal estimate is added to the quantized difference signal to produce the reconstructed version of the input signal.
- Both the reconstructed signal and the quantized difference signal are operated upon by an adaptive predictor which produces the input signal estimate, thus forming a feedback loop.
- the embodiment of the invention herein described is concerned primarily with the adaptive predictor portion of the ADPCM encoder, and in particular the filtering operation of the adaptive predictor.
- the filtering operation of the adaptive predictor is the most complex block of the ADPCM algorithm. According to the ITU-T recommendation, this involves first convening the fixed point inputs to floating point, multiplying the mantissas and adding the exponents, and finally converting the floating point product back to fixed point representation.
- the values d q and s t are close to the 16-bit limit, converting them to floating point will result in a loss of precision even before the multiply operation can take place. This is because only 6 bits are retained for the mantissa of the floating point number.
- the preferred embodiment of the present invention provides a way in which to perform the operations more efficiently and accurately.
- the preferred embodiment of the invention primarily involves four function blocks that are defined in ITU-T Recommendation G.726, namely the FLOATA, FLOATB, FMULT and ACCUM blocks.
- the functions of these blocks as utilised in the ITU-T recommendation are described briefly below with reference to the figure and the signal estimate equation mentioned above.
- FLOATA This block receives the quantized difference signal d q as input, where the quantity d q is defined as a 15 or 16 bit signed binary magnitude.
- the quantized difference signal d q is converted into a floating point value. This is performed by computing the exponent and mantissa and combining the sign bit, 4 exponent bits and 6 mantissa bits into one 11 bit word.
- FLOATB This block receives the reconstructed signal s t as input, where the quantity s t is defined as a 16 bit twos-complement quantity.
- the reconstructed signal s ⁇ is converted into a floating point value. This is performed by computing the exponent and mantissa and combining the sign bit, 4 exponent bits and 6 mantissa bits into one 11 bit word.
- FMULT This block multiplies predictor coefficients with the corresponding quantized difference signal or reconstructed signal.
- the multiplication is done in a floating point format, and thus the predictor coefficients, which are defined as 16 bit twos-complement quantities, are first converted into floating point representations.
- the products of the multiplication operations are signal estimate partial products (WAn, WBn), which are also defined as 16 bit twos- complement quantities, requiring a conversion from the floating point multiplication result.
- ACCUM This block operates on the signal estimate partial products to perform the summing portion of the operation represented by the equation discussed above.
- the partial products of the signal estimates (WA1, WA2, WB1, WB2, WB3, WB4, WB5 and WB6) are received as input and summed to obtain the complete signal estimate s e . All of the quantities are twos-complement representations.
- the preferred embodiment employs fixed point multiplication rather than floating point computation, which requires a number of modifications as described below.
- Replacing the floating point multiplication with a fixed point multiplication eliminates the need to convert values between fixed and floating point formats. This significantly reduces the complexity of the overall algorithm. By omitting the fixed-to-floating point conversion, the full precision of the original values is preserved, thereby reducing errors due to loss of precision. As a result, an improvement in the quality of the decoded signal can be achieved.
- the primary modifications to the ITU-T recommended ADPCM adaptive predictive filter which are implemented in the preferred embodiment are summarised below.
- the FLOATA block ordinarily takes the reconstructed signal and converts it from 16-bit signed magnitude format to floating point. This block is re-defined to convert the signed magnitude numbers to 16-bit twos-complement numbers instead.
- the FLOATB block ordinarily takes the signal estimate and converts it from 16-bit twos-complement to floating point. This block is no longer needed and is discarded from the system.
- the FMULT block normally performs several functions, namely converting the predictor filter coefficients from 16-bit twos-complement to floating point, performing floating point multiplication by adding the exponents and multiplying the mantissas and finally converting the product bxk into a 16-bit twos-complement number.
- the preferred embodiment requires that all these functions be discarded and replaced by a simple fixed-point multiplication which multiplies two 16-bit twos-complement numbers to give a 32-bit product. The full 32 bits of the result is retained. No truncation to 16 bits is performed.
- the ACCUM block ordinarily adds the 16-bit predictor outputs together to form the signal estimate.
- the preferred embodiment requires that the accumulation function be modified to operate on 32-bit inputs. After the final accumulation, the result is then rounded off to 16 bits to give the signal estimate.
- the preferred embodiment of the invention requires the ability to perform 16x16 bit fixed-point multiplication and to store the 32-bit result for subsequent arithmetic operations.
- the following procedures, in pseudocode, implement the preferred embodiment by redefining the FLOATA, FLOATB, FMULT and ACCUM blocks originally specified in ITU-T G.726.
- the modified procedures in combination retain the functionality of the ITU-T recommendation, although not in strict compliance with the specification.
- Table 1, below, provides a description of the format of the variables used in the procedures.
- TC denotes twos-complement representation
- SM denotes signed magnitude representation denotes sign bit
- DQS DQ > > 15 Get the sign bit.
- WAn An x SRn
- WBn Bn x DQn
- Procedure 4 ACCUM: Function: Addition of predictor outputs to f ⁇ rm the partial signal estimate (from the sixth order predictor) and the signal estimate.
Abstract
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Priority Applications (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
EP98959358A EP1138037A1 (en) | 1998-12-02 | 1998-12-02 | Fixed-point multiplication for adpcm speech coder |
PCT/SG1998/000098 WO2000033293A1 (en) | 1998-12-02 | 1998-12-02 | Fixed-point multiplication for adpcm speech coder |
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
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PCT/SG1998/000098 WO2000033293A1 (en) | 1998-12-02 | 1998-12-02 | Fixed-point multiplication for adpcm speech coder |
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WO2000033293A1 true WO2000033293A1 (en) | 2000-06-08 |
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PCT/SG1998/000098 WO2000033293A1 (en) | 1998-12-02 | 1998-12-02 | Fixed-point multiplication for adpcm speech coder |
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WO (1) | WO2000033293A1 (en) |
Citations (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4876660A (en) * | 1987-03-20 | 1989-10-24 | Bipolar Integrated Technology, Inc. | Fixed-point multiplier-accumulator architecture |
-
1998
- 1998-12-02 EP EP98959358A patent/EP1138037A1/en not_active Withdrawn
- 1998-12-02 WO PCT/SG1998/000098 patent/WO2000033293A1/en not_active Application Discontinuation
Patent Citations (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4876660A (en) * | 1987-03-20 | 1989-10-24 | Bipolar Integrated Technology, Inc. | Fixed-point multiplier-accumulator architecture |
Non-Patent Citations (1)
Title |
---|
TOSHIHIKO MATSUMURA ET AL: "VLSI DSP (MB8764) BASED ADAPTIVE SPEECH CODEC", ION EXCHANGE, vol. PART 14, no. VOL. 50, 5 June 1985 (1985-06-05), ANTWERPEN F J VAN, pages 1121 - 1124, XP000012513 * |
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