CN109416913B - Adaptive audio coding and decoding system, method, device and medium - Google Patents

Adaptive audio coding and decoding system, method, device and medium Download PDF

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CN109416913B
CN109416913B CN201780040686.9A CN201780040686A CN109416913B CN 109416913 B CN109416913 B CN 109416913B CN 201780040686 A CN201780040686 A CN 201780040686A CN 109416913 B CN109416913 B CN 109416913B
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filter
signal
low pass
pass filter
step size
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CN109416913A (en
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J·约翰斯顿
S·怀特
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Immersion Services LLC
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Immersion Services LLC
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Priority claimed from US15/151,109 external-priority patent/US10699725B2/en
Priority claimed from US15/151,220 external-priority patent/US10756755B2/en
Priority claimed from US15/151,200 external-priority patent/US10770088B2/en
Priority claimed from US15/151,211 external-priority patent/US20170330575A1/en
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    • 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
    • G10L19/26—Pre-filtering or post-filtering
    • G10L19/265—Pre-filtering, e.g. high frequency emphasis prior to encoding
    • 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
    • G10L19/032—Quantisation or dequantisation of spectral components
    • G10L19/035—Scalar quantisation
    • 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
    • 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
    • G10L19/26—Pre-filtering or post-filtering
    • 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
    • G10L2019/0001—Codebooks

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  • 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)

Abstract

An encoder comprising a low pass filter for filtering an input audio signal. The low pass filter has fixed filter coefficients. The encoder generates a quantized signal based on the difference signal. The encoder includes an adaptive quantizer and a decoder to generate a feedback signal. The decoder has an inverse quantizer and a predictor. The predictor has a fixed control parameter based on the frequency response of the low pass filter. The predictor may include a finite impulse response filter having fixed filter coefficients. The decoder may include an adaptive noise shaping filter coupled between the low pass filter and the encoder. The adaptive noise shaping filter flattens signals within a spectrum corresponding to the spectrum of the low pass filter.

Description

Adaptive audio coding and decoding system, method, device and medium
Cross Reference to Related Applications
This patent application claims priority from U.S. patent application Ser. No. 15/151,109, U.S. patent application Ser. No. 15/151,200, U.S. patent application Ser. No. 15/151,211, and U.S. patent application Ser. No. 15/151,220, all of which are entitled "adaptive Audio codec systems, methods, and products," and all of which were filed on U.S. patent and trademark office at 5 and 10 of 2016.
Technical Field
The present specification relates to systems, methods, and products for encoding and decoding audio signals.
Background
Differential Pulse Code Modulation (DPCM) may be used to reduce the noise level or bit rate of an audio signal. The difference between the input audio signal and the predicted signal may be quantized to generate an output encoded data stream with reduced energy. The prediction signal of the encoder may be generated using a decoder including an inverse quantizer and predictor circuit. An Adaptive Differential Pulse Code Modulation (ADPCM) changes the quantization step size of a quantizer (and an inverse quantizer) to improve efficiency in consideration of a varying dynamic range of an input signal.
Disclosure of Invention
In one embodiment, an apparatus comprises: a low pass filter having determined filter coefficients and configured to filter an input signal; an encoder configured to generate a quantized signal based on a difference signal and comprising: an adaptive quantizer; and a decoder configured to generate a feedback signal and having an inverse quantizer and predictor circuit having determined control parameters based on a frequency response of the low pass filter. In one embodiment, the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter, the predictor circuit comprises a Finite Impulse Response (FIR) filter, and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter. In one embodiment, the apparatus comprises: an adaptive noise shaping filter coupled between the low pass filter and the encoder, the adaptive noise shaping filter configured to flatten signals within a spectrum corresponding to a spectrum of the low pass filter. . In one embodiment, the adaptive noise shaping filter is configured not to flatten frequencies above the edge frequency of the low pass filter. . In one embodiment, the edge frequency is 25kHz. In one embodiment, the adaptive noise shaping filter generates a signal indicative of filter coefficients of the adaptive noise shaping filter, the signal indicative of filter coefficients of the adaptive noise shaping filter being included in a bitstream output by the encoder. In one embodiment, the encoder includes an encoder circuit configured to generate a codeword based on a quantized signal word generated by the adaptive quantizer. In one embodiment, the encoder circuit is configured to generate the escape code in response to at least one of; quantized signal words not associated with a respective encoded codeword; ending of the signal path of the signal to be encoded; and an end of the signal to be encoded. In an embodiment, the encoder circuit is configured to generate the codeword using huffman coding. In one embodiment, the adaptive quantizer is a variable rate quantizer. In one embodiment, the step size and bit rate of the quantized signal generated by the adaptive quantizer is variable. In one embodiment, the adaptive quantizer is configured to control the step size according to the following equation:
d n +1＝βd n +m(c n /L factor )，
Wherein c n Is the current quantized signal word, d n Corresponding to the current step in the logarithmic domain, L factor Is the load factor, m (c) n /L factor ) Is based on the current quantized signal c n And the load factor L factor A selected logarithmic multiplier, β is the leakage coefficient, and d n+1 Corresponding to the next quantized signal word c to be applied n+1 Is used for the step size in the logarithmic domain. In one embodiment, the adaptive quantizer is configured to control the step size according to the following equation:
d n +1＝max(βd n +m(c n /Lf actor ),d min )，
wherein c n Is the current quantized signal word, d n Corresponding to the current step in the logarithmic domain, L factor Is the load factor, m (c) n /L factor ) Is based on the current quantized signal c n And the load factor L factor The selected logarithmic multiplier, β is the leakage coefficient, d min Is the threshold step in the logarithmic domain, and d n+1 Corresponding to the next quantized signal word c to be applied n+1 Is used for the step size in the logarithmic domain.
In one embodiment, a method includes: filtering the input signal, the filtering the input signal comprising using a low pass filter having determined filter coefficients; and encoding the filtered input signal using a feedback loop, the encoding the filtered input signal comprising: generating a quantized signal based on the difference signal using an adaptive quantizer; generating a feedback signal based on the quantized signal using an inverse quantizer and a predictor circuit, the predictor circuit having determined control parameters based on a frequency response of the low pass filter; and generating the difference signal based on the feedback signal and the filtered input signal. In one embodiment, the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter, the predictor circuit comprises a Finite Impulse Response (FIR) filter, and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter. In one embodiment, the filtering the input signal includes filtering the signal output by the low pass filter using an adaptive noise shaping filter that flattens the signal within a spectrum corresponding to the spectrum of the low pass filter. In one embodiment, the method comprises: a signal indicative of the filter coefficients of the adaptive noise shaping filter is generated and included in the encoded bitstream. In one embodiment, the method comprises: a codeword is generated based on the quantized signal word generated by the adaptive quantizer. In one embodiment, the method comprises: generating an escape code in response to at least one of: quantized signal words associated with respective encoded codewords; ending of the signal path of the signal to be encoded; and an end of the signal to be encoded. In one embodiment, the method comprises: controlling the step size of the adaptive quantizer according to the following equation:
d n+1 ＝max(βd n +m(c n /Lf actor ),d min )，
Wherein c n Is the current quantized signal word, d n Corresponding to the current step in the logarithmic domain, L factor Is the load factor, m (c) n /L factor ) Is based on the current quantized signal c n And the load factor L factor The selected logarithmic multiplier, β is the leakage coefficient, d min Is the threshold step in the logarithmic domain, and d n+1 Corresponding to the next quantized signal word c to be applied n+1 Is used for the step size in the logarithmic domain.
In one embodiment, the contents of a non-transitory computer readable medium configure a signal processing circuit to perform a method comprising: filtering the input signal, the filtering the input signal comprising using a low pass filter having determined filter coefficients; and encoding the filtered input signal using a feedback loop, the encoding the filtered input signal comprising: generating a quantized signal based on the difference signal; generating a feedback signal based on the quantized signal using determined control parameters based on a frequency response of the low pass filter; and generating the difference signal based on the feedback signal and the filtered input signal. In one embodiment, the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter, generating the feedback signal comprises using a Finite Impulse Response (FIR) filter, and the determined control parameters comprise fixed filter coefficients of the FIR filter. In one embodiment, the filtering the input signal includes adaptive noise shaping to flatten the signal within a spectrum corresponding to the spectrum of the low pass filter. In one embodiment, the method comprises: controlling a step size of generating the quantized signal according to the following equation:
d n+1 ＝max(βd n +m(c n /Lf actor ),d min )，
Wherein c n Is the current quantized signal word, d n Corresponding to the current step in the logarithmic domain, L factor Is the load factor, m (c) n /L factor ) Is based on the current quantized signal c n And the load factor L factor The selected logarithmic multiplier, β is the leakage coefficient, d min Is the threshold step in the logarithmic domain, and d n+1 Corresponding to the next quantized signal word c to be applied n+1 Is used for the step size in the logarithmic domain.
In one embodiment, a system includes: an encoder, the encoder comprising: a low pass filter having determined filter coefficients and configured to filter an input signal; an adaptive quantizer configured to generate a quantized signal based on the difference signal; an inverse quantizer; and a predictor circuit coupled between the adaptive quantizer and the predictor circuit, the predictor circuit having determined control parameters based on a frequency response of the low pass filter; and a decoder configured to decode the signal encoded by the encoder. In one embodiment, the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter, the predictor circuit comprises a Finite Impulse Response (FIR) filter, and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter. In one embodiment, the system comprises: an adaptive noise shaping filter is coupled between the low pass filter and the adaptive quantizer, the adaptive noise shaping filter configured to flatten signals within a spectrum corresponding to a spectrum of the low pass filter. In one embodiment, the adaptive noise shaping filter generates a signal indicative of filter coefficients of the adaptive noise shaping filter, the signal indicative of filter coefficients of the adaptive noise shaping filter being included in a bitstream output by the encoder to the decoder. In one embodiment, the encoder comprises encoder circuitry configured to generate codewords based on quantized signal words generated by the adaptive quantizer, the decoder comprising decoding circuitry configured to generate quantized signal words based on codewords generated by the encoder circuitry. In one embodiment, the encoder circuit and the decoding circuit are configured to use escape encoding.
In one embodiment, a system includes: an input filter having determined control parameters and configured to limit a bandwidth of an input signal to less than seventy-five percent of an available bandwidth based on a sampling frequency of the input signal; an encoder configured to generate a quantized signal based on a difference signal, and comprising: an adaptive quantizer; and a feedback circuit configured to generate a feedback signal and having an inverse quantizer and predictor circuit having determined control parameters based on a frequency response of the input filter. In one embodiment, the system comprises: a decoder configured to decode the signal encoded by the encoder. In one embodiment, the input filter is a low pass filter, the determined control parameter of the low pass filter is a fixed filter coefficient of the low pass filter, the predictor circuit comprises a Finite Impulse Response (FIR) filter, and the determined control parameter of the predictor circuit comprises a fixed filter coefficient of the FIR filter. In one embodiment, the input filter is a bandpass filter, the determined control parameter of the bandpass filter is a fixed filter coefficient of the bandpass filter, the predictor circuit comprises a Finite Impulse Response (FIR) filter, and the determined control parameter of the predictor circuit comprises a fixed filter coefficient of the FIR filter.
In one embodiment, a system includes: low-pass filtering means for low-pass filtering the input signal using the determined filtering parameters; quantized signal generating means for generating a quantized signal based on a difference signal; prediction signal generation means for generating a prediction signal based on the quantized signal using determined control parameters based on the frequency response of the low pass filtering means; and a difference signal generating means for generating the difference signal. In one embodiment, the system comprises: decoding means for decoding the encoded signal. In one embodiment, the low-pass filtering means comprises a low-pass filter having fixed filter coefficients, and the prediction signal generating means comprises a Finite Impulse Response (FIR) filter having fixed filter coefficients based on the filter coefficients of the low-pass filter.
Drawings
Fig. 1 is a functional block diagram of an embodiment of an ADPCM encoder.
Fig. 2 is a functional block diagram of an embodiment of an ADPCM decoder.
Fig. 3 is a functional block diagram of an embodiment of a quantizer step size control circuit.
Fig. 4 is a functional block diagram of an embodiment of an ADPCM encoder.
Fig. 5 shows an exemplary frequency response of an embodiment of a low pass filter.
Fig. 6 shows an embodiment of a method of controlling a change in an adaptive quantizer step size.
Fig. 7 is a functional block diagram of an embodiment of an ADPCM decoder.
Fig. 8 is a functional block diagram of an embodiment of a quantizer step size and bit rate control circuit.
Fig. 9 shows an embodiment of a method of generating codewords and controlling the variation of the adaptive quantizer step size.
Fig. 10 illustrates an embodiment of a method of generating quantized signal values from codewords.
Detailed Description
In the following description, certain details are set forth in order to provide a thorough understanding of various embodiments of the apparatus, system, method, and article. However, it will be understood by those skilled in the art that other embodiments may be practiced without these details. In other instances, well-known structures and methods, such as transistors, multipliers, integrated circuits, etc., associated with, for example, impulse response filters, encoders, decoders, audio and digital signal processing circuits, etc., have not been shown or described in detail in some figures in order to avoid unnecessarily obscuring descriptions of the embodiments.
Throughout the specification and the claims which follow, unless the context requires otherwise, the word "comprise", and variations such as "comprises" and "comprising", will be interpreted in an open, inclusive sense, i.e. "including but not limited to. "
Reference throughout this specification to "one embodiment" or "an embodiment" means that a particular feature, structure, or characteristic described in connection with the embodiment is included in at least one embodiment. Thus, the appearances of the phrases "in one embodiment" or "in an embodiment" in various places throughout this specification are not necessarily all referring to the same embodiment, nor are all embodiments. Furthermore, the particular features, structures, or characteristics may be combined in any suitable manner in one or more embodiments to obtain further embodiments.
Headings are provided for convenience only and do not interpret the scope or meaning of the present disclosure.
The dimensions and relative positioning of elements in the drawings are not necessarily drawn to scale. For example, the shapes of various elements and angles are not drawn to scale, and some of these elements are exaggerated and positioned to improve legibility of the drawing figs. Furthermore, the particular shapes of the elements as drawn, are not necessarily intended to convey any information regarding the actual shape of the particular elements, and have been solely selected for ease of recognition in the drawings.
Fig. 1 is a functional block diagram of an embodiment of an audio signal encoder 100 that may employ Adaptive Differential Pulse Code Modulation (ADPCM). As shown in fig. 1, encoder 100 has adder circuit 110, adaptive quantizer circuit 120, decoder circuit 130 including inverse quantizer circuit 134 and predictor circuit 138, quantizer step size control circuit 140, and optional encoder circuit 150.
In operation of the embodiment, an analog input audio signal to be encoded is received at the positive input 112 of the adder 110 of the encoder 100. The negative input 114 of the adder 110 receives as a feedback signal the prediction signal generated by the decoder 130. Adder 110 generates a difference signal that is provided to adaptive quantizer circuit 120. The adaptive quantizer circuit 120 may be an analog-to-digital converter that samples the received difference signal and generates an output signal representative of the difference signal as a series of quantized signals representative of different signal levels. For example, an 8-bit word may be used to represent 256 different signal levels (e.g., 256 different steps with uniform steps); a 4-bit word may be used to represent 16 different signal levels; etc. Alternatively, in one embodiment, the quantized signal may be encoded by encoder circuit 150, such as huffman encoding and/or arithmetic encoding, to generate an encoded signal output. The quantized signal output by the adaptive quantizer circuit 120 (or the output of the optional encoder 150 when an encoder is employed) is the output quantized signal or codeword of the encoder 100. The quantizer step size control circuit 140 generates control signals to control the size of the quantization step sizes employed by the quantizer 120 (and the inverse quantizer 134), which may be varied to facilitate efficient transmission, storage, etc., in view of input audio signals having varying dynamic ranges.
The inverse quantizer 134 of the decoder 130 generates a signal such as an analog signal based on the quantized signal output by the adaptive 25 quantizer and the current step size control signal set by the quantizer step size control circuit 140. The predictor circuit 138 may generate a prediction signal based on the output signal of the inverse quantizer 134 and historical data, such as the most recently quantized signal values and the most recently predicted signal values. The predictor circuit 138 may employ one or more filters and one or more feedback loops.
As shown, the encoder 100 of fig. 1 includes one or more processors or processor cores P, one or more memories M, and discrete circuits DC, which may be used alone or in various combinations to implement the functions of the encoder 100. In operation, an embodiment of the encoder 100 generates quantized and optionally encoded data from an input analog audio signal. In operation of embodiments, a digital audio signal to be encoded (e.g., to be encoded as a reduced bit stream) rather than an analog signal may be received at the positive input 112 (e.g., an 8-bit digital audio signal may be encoded as a 4-bit digital audio signal).
Although the components of encoder 100 of fig. 1 are shown as separate components, the various components may be combined (e.g., quantizer step size control circuit 140 may be integrated into adaptive quantizer 120 in some embodiments) or separated into additional components (e.g., predictor circuit 138 may be separated into multiple predictor circuits, separated components such as filters, adders, buffers, look-up tables, etc.), and various combinations thereof.
Fig. 2 is a functional block diagram of an embodiment of an audio signal decoder 200 that may employ Adaptive Differential Pulse Code Modulation (ADPCM). Decoder 200 may be used, for example, as decoder 130 of fig. 1; acting as a separate decoder to decode the received encoded signal; etc. As shown in fig. 2, decoder 200 has optional decoding circuitry 250, inverse quantizer circuitry 234, predictor circuitry 238, inverse quantizer step size control circuitry 240, and adder 270.
In operation of an embodiment, an encoded signal is received by decoding circuit 250, which decoding circuit 250 converts the encoded signal to a quantized signal. The quantized signal to be decoded is provided to an inverse quantizer 234 and to an inverse quantizer step size control circuit 240. When decoder 200 is used in an encoder (e.g., encoder 100 of fig. 1), decoding circuit 250 may generally be omitted and the same step size control circuit may be used to provide step size control signals to the quantizer as well as to the inverse quantizer (see fig. 1). The inverse quantizer 234 generates a signal such as an analog signal based on the quantized signal output by the decoding circuit 250 (or received from the quantizer (see the quantizer 120 of fig. 1)) and the current step size set by the inverse quantizer step size control circuit 240. The output of the inverse quantizer 234 is provided to a first positive input of an adder 270. The output of the adder is provided to a predictor 238, which as shown includes a Finite Impulse Response (FIR) filter. The output of the FIR filter is provided to a second positive input of adder 270.
When the decoder 200 is used as a decoder to provide a decoded signal as an output, the output of the decoder 200 is the output of the adder 270. When decoder 200 is used as part of a feedback loop in an encoder, such as decoder 130 used in encoder 100 of fig. 1, the output of predictor circuit 238 provides a prediction signal to the encoder (see the prediction signal provided to negative input 114 of adder 110 of fig. 1).
The inverse quantizer 234, the inverse quantizer step size control circuit 240, and the predictor circuit 238 are generally capable of operating in a similar manner as corresponding components of an encoder, such as the encoder 100 of fig. 1. For example, referring to fig. 1 and 2, having the corresponding components in the encoder 100 and decoder 200 operate in a similar manner facilitates generating a prediction signal using the quantized signal and controlling the step sizes in the encoder 100 and decoder 200 without exchanging additional control signals between the encoder 100 and decoder 200.
As shown, the decoder 200 of fig. 2 includes one or more processors or processor cores P, one or more memories M, and discrete circuits DC, which may be used alone or in various combinations to implement the functions of the decoder 200. Although the components of decoder 200 of fig. 2 are shown as separate components, the various components may be combined (e.g., in some embodiments, inverse quantizer step size control circuit 240 may be integrated into inverse quantizer 234) or separated into additional components (e.g., predictor circuit 238 may be separated into separate components, such as filters, adders, buffers, look-up tables, etc.), and various combinations thereof.
Fig. 3 is a functional block diagram of an embodiment of a quantizer step size control circuit 340 that may be used, for example, as the quantizer step size control circuit 140 in the embodiment of the encoder 100 of fig. 1, or as the inverse quantizer step size control circuit 140 in the embodiment of the decoder 200 of fig. 2. As shown, the quantizer step size control circuit 340 includes a logarithmic multiplier selector 342 that selects a logarithmic multiplier based on the current quantized signal word (as shown, the word output by the adaptive quantizer 320). In some embodiments, the current quantized signal word may be included in a bitstream being decoded by a decoder (see fig. 2). The logarithmic multiplier selector 342 may select the logarithmic multiplier based on historical data (e.g., previous quantized signal words) and may include a look-up table LUT, such as: updating based on historical data can be performed on the update download, and the like. The logarithmic multiplier selector 342 may select a logarithmic multiplier based on statistical probabilities based on current and previous quantized signal words. Quantizer step size control circuit 340 includes adder 344 that receives the selected logarithmic multiplier at a first positive input and provides an output to delay circuit 346. The output of delay circuit 346 is provided to multiplier 348 and to exponent circuit 350. Multiplier 348 multiplies the output of delay circuit 346 by a scale or leakage factor β, which may generally be close to and less than 1, and provides the result to a second positive input of adder 344. The leakage factor may generally be constant, but may be variable in some embodiments, for example, based on previous step control signals or other historical data. Since the introduced error will decay, selecting the scaling factor β close to and less than 1 is beneficial in reducing the impact of selecting an incorrect step size, e.g., due to transmission errors.
In operation, the exponent circuit 350 generates a step control signal based on the output of the delay circuit 346. As shown, the step control signal is provided to the adaptive quantizer 320 and to the inverse quantizer 334. As shown. The quantizer step size control circuit 340 operates in a logarithmic manner, which may simplify the calculation. Some embodiments may operate in a linear fashion and may employ, for example, multipliers instead of adders 244 and exponential circuits instead of multipliers 246. The quantizer step size control circuit 340 as shown operates logarithmically and the step size selected based on the step size control signal varies exponentially.
In one embodiment, quantizer step size control circuit 340 may operate according to equation 1 below:
d n+1 ＝βd n +m(c n ) Equation 1
Wherein d n Is the step in the logarithmic domain, m (c n ) Is the logarithmic multiplier selected based on the current quantized signal, β is the scale factor or leakage coefficient. As shown, fig. 3 includes one or more processors P, one or more memories M, and discrete circuits DC, which may be used alone or in various combinations to implement the functions of quantizer step size control circuit 340.
Although the components of fig. 3 are shown as separate components, the various components may be combined (e.g., adder 344 and multiplier 348 may be integrated into an arithmetic processor in some embodiments) or separated into additional components and various combinations thereof.
Fig. 4 is a functional block diagram of an audio signal encoder 400 that may employ Adaptive Differential Pulse Code Modulation (ADPCM). The audio signal encoder 400 of an embodiment provides additional bandwidth control, helps to avoid quantizer overload, and includes adaptive noise shaping. As shown in fig. 4, encoder 400 has a low pass filter 475, an adaptive noise shaping filter 480, an adder circuit 410, a variable rate adaptive quantizer circuit 420, a decoder circuit 430 including an inverse quantizer circuit 434 and a predictor circuit 438, a quantizer step size and average bit rate control circuit 440, an encoder 450, and a bit stream assembler 485.
In operation of the embodiment, an analog input audio signal to be encoded is received at the input of the input filter, as shown by low pass filter 475. The low pass filter 475 helps improve the signal-to-noise ratio. The low pass filter 475 may be, for example, a FIR filter with 25kHz edges and 30kHz stop band, which has been found to provide excellent results for data sampled at 88.2 or 96 kHz. Fig. 5 shows an exemplary frequency response of an embodiment of the low pass filter 475 applied to a 96kHz sampling rate. When a sufficiently high sampling rate is employed, the use of a low pass filter and a corresponding fixed predictor filter (e.g., the predictor employing filter coefficients based on the frequency response of the input filter) with the employed control parameters based on the control parameters of the input filter helps to obtain a substantial prediction gain for the input signal, which in turn helps to obtain the required minimum signal-to-noise ratio. In testing, sample rates below 48kHz (e.g., 44.1 and 48 kHz) generally do not provide adequate improvement in gain.
The output of the low pass filter 475 is provided to an adaptive noise shaping filter 480. In some embodiments, the low pass filter 475 may be omitted and the signal to be encoded may be input to the adaptive noise shaping filter 480 instead of the low pass filter 475. In some embodiments, the adaptive noise shaping filter 480 may be omitted or selectively bypassed. For example, when encoding with a high bit rate signal, the adaptive noise shaping filter 480 may be omitted or bypassed. In some embodiments, a band pass filter may be used instead of the low pass filter and the predictor filter adjusted accordingly. For example, in one embodiment, an input filter (e.g., a band-pass filter) having fixed control parameters and configured to limit the bandwidth of the input signal to less than seventy-five percent of the available bandwidth based on the sampling frequency may be employed, and the corresponding decoder may include a predictor circuit having fixed control parameters based on the frequency response of the filter. The use of an input filter to limit the bandwidth of the input signal and to set control parameters of the predictor circuit based on the frequency response of the input filter helps to obtain a substantial prediction gain of the input signal when a sufficiently high sampling rate is employed, which in turn helps to obtain the required minimum signal-to-noise ratio.
The adaptive noise shaping filter 480 may be, for example, a low order all zero linear prediction filter. Real (non-complex) coefficients may be employed. In one embodiment, the adaptive noise shaping filter 480 is an all-zero adaptive noise shaping filter that flattens the spectrum of the signal received from the low pass filter 475 while maintaining an overall spectral slope and sufficient masking to maintain transparent codec (e.g., compression artifacts are typically imperceptible). In a corresponding decoder (see decoder 700 of fig. 7), an all-pole filter using the same coefficients may be used to restore the original spectral shape. In one embodiment, adaptive noise shaping filter 480 retains whiteness criteria for predictor circuit 438. For example, the low-order noise shaping filter 480 may be adjusted so as not to planarize the signal at the edge frequency of the low-pass filter (e.g., 25kHz, which may not be present in the signal filtered by the low-pass filter 475). As described above, the missing energy at high frequency contributes to improvement of the prediction gain. Filters other than linear prediction filters may be used as noise shaping filters.
The adaptive noise shaping filter 480 provides the filtered output signal to the positive input 412 of the adder 410. In one embodiment, the adaptive noise shaping filter 480 also provides a signal including adaptive noise filter setting information and/or synchronization information that may be used to communicate the adaptive noise filter setting and synchronization information to a decoder, such as the decoder 700 of fig. 7, that includes a corresponding inverse noise shaping filter 780. The setting and synchronization information may be sent periodically, for example once every 512 sample blocks. In some embodiments, the adaptive noise shaping filter control information may be implicit in the codewords of the bitstream. For example, when a codeword of the bitstream indicates that an average bit rate above a threshold average bit rate is being employed, this may also indicate that adaptive noise shaping is being bypassed.
The negative input 414 of adder 410 receives as a feedback signal the prediction signal generated by decoder 430. Adder 410 generates a difference signal that is provided to variable rate adaptive quantizer circuit 420.
The variable rate adaptive quantizer circuit 420 generates an output signal representing the difference signal as a series of quantized signals or words. As discussed in more detail below, the size of the quantized signal is not fixed and the average length may be adjusted using the step size and the output of the multiplier table of the average bit rate controller 440. The output of the variable rate adaptive quantizer circuit 420 is provided to a step size and average bit rate controller 440. An inverse quantizer 434 and an encoder 450.
The quantizer step size and average bit rate control circuit 440 generates one or more control signals to control the size of the quantization step size. This implicitly determines the average length of the quantized signal employed by quantizer 420 (and inverse quantizer 434), which may be changed by adjusting the multiplier table in order to efficiently encode, given that the input audio signal has a varying dynamic range.
Fig. 6 illustrates an embodiment of a method 600 of generating codewords and controlling the variation of step size and average bit rate that may be employed, for example, by the encoder 400 of fig. 4. For convenience, the method 600 will be described with reference to the encoder 400 of fig. 4. The method begins at 602 and proceeds to 604. At 604, the variable rate adaptive quantizer 420 generates a current quantized signal or word based on the difference signal and the current quantization step size control signal. This may be done, for example, according to equation 2 below:
Wherein c n Is the current quantized signal, e n Is an error or difference signal, d n Corresponding to the current step in the log domain.
The method proceeds from 604 to 606. At 606, the quantizer step size and average bit rate control circuit 440 generates one or more control signals to set the step size of the next quantized signal word. This may be done, for example, according to equation 1 above or according to equations 3 or 4 below:
d n+1 ＝βd n +m(c n /L factor ) Equation 3
Wherein c n Is the current quantized signal, d n Corresponding to the current step size and in response to the bit length, L factor Is the load factor for controlling the average bit length (and thus the average bit rate), m (c) n /L factor ) Is the logarithmic multiplier selected based on the current quantized signal and the load factor, β is the leakage factor. In some embodiments, a minimum step d in the logarithmic domain may be set min The following are provided:
d n+1 ＝max(βd n +m(c n /L factor ),d min ) Equation 4
The load factor L can be selected factor In order to maintain a desired average bit rate. The load factor may typically be between 0.5 and 16. In some embodiments, a maximum step size may be employed. Changing the logarithmic multiplier m (c) n /L factor ) To change the bit rate and step size and may select the values stored in the look-up table of the logarithmic multiplier selector (see fig. 8) to cause the adaptive quantizer 420 and the inverse quantizer 434 to achieve the desired change in step size and bit rate. For example, a larger logarithmic multiplier may indicate an increased step size and a lower bit rate by the quantizer 420 and the inverse quantizer 434. Can be based on the current quantized value c n Divided by load factor L factor Is used to index the look-up table. In addition to L factor In addition, different look-up tables may be employed instead of or in addition to different load factors. In one embodiment, the values in the lookup table may be selected such that the logarithmic multiplier follows the current quantized value c n Monotonically increasing from an increase in (c), and the multiplier table may be increased from a small c of negative value n Large c becoming positive n 。
Method 600 proceeds from 606 to 608. At 608, the encoder 400 determines whether to continue encoding the received signal. When it is determined at 608 to continue encoding the received signal, the method returns to 604 to process the next quantized signal word. When it is determined at 608 that the received signal is not to be encoded, the method proceeds to 610, where other processing may occur, such as: an escape code is generated to indicate that the received signal has terminated, and so on. The method proceeds from 610 to 612 where the method 600 terminates at 612.
Some embodiments of encoder 400 may perform other acts not shown in fig. 6, may perform all acts not shown in fig. 6, or may perform the acts of fig. 6 in a different order.
Referring to fig. 4, the inverse quantizer 434 of the decoder 430 outputs c based on the quantized signal of the variable rate adaptive quantizer 420 n And the current step size d n A signal such as an analog signal is generated. As discussed in more detail below with reference to fig. 7, the predictor circuit 438 may generate a prediction signal based on the output signal of the inverse quantizer 434 and historical data, such as the most recently encoded data and the most recently predicted values. As discussed in more detail below with reference to fig. 7, the predictor circuit 438 may employ a FIR filter having coefficients selected based on the frequency response of the low pass filter 475. These coefficients may be fixed and may be selected so as to maintain a sufficient signal-to-noise ratio for the desired input signal characteristics. Tests have shown that the use of fixed coefficients based on the frequency response of the low pass filter 475 for the FIR filter in the predictor circuit 438 results in a significant improvement in the signal-to-noise ratio of signals at 64kHz and above. For example, in one embodiment, attenuating energy above 25kHz in the low pass filter 475 and selecting fixed coefficients of the FIR filter based on the frequency response of the low pass filter may result in a prediction gain of 45 dB. The use of an eight-bit quantizer (see adaptive quantizer 120 of fig. 1, which may be an eight-bit quantizer, a four-bit quantizer, etc.) may result in a signal-to-noise ratio comparable to encoding without the use of an adaptive noise shaping filter (see fig. 1), but does not include frequencies above 25 kHz.
In one embodiment, the quantized signal output by the variable rate adaptive quantizer circuit 420 (or, when an encoder is employed, the optional encoder 450) is the output quantized signal of the encoder 400. Alternatively, in one embodiment, the quantized signal may be encoded by encoder circuit 450 using, for example, huffman encoding and/or arithmetic encoding to generate the encoded signal output of encoder 400. Encoder 450 converts the quantized signal words into codewords, for example, using one or more look-up tables. Less used quantized signal words may be assigned to larger codewords and more frequently used quantized signal words may be assigned to smaller codewords to increase the efficiency of encoder 400.
In one embodiment, encoder 450 optionally provides escape encoding. For example, for quantized values not included in the employed codebook (e.g., huffman codebook), an escape code may be transmitted instead of codewords from the codebook, where the escape code indicates how quantized signal values or information will be transmitted (e.g., actual quantized signal is being transmitted; next codeword is quantized signal value instead of codeword; difference between maximum/minimum levels is being transmitted; etc.). In another example, the escape code may indicate that a channel of the encoded signal is being interrupted or is not present (e.g., only one channel of the stereo signal is being encoded). In another example, the escape code may indicate the end of the encoded signal.
The bit stream assembler 485 receives the codewords output by the encoder 450 and the adaptive noise shaping filter control/synchronization information output by the adaptive noise shaping filter 480 and assembles the bit stream for transmission to a decoder and/or memory. In some embodiments, the data packets may be assembled by a bit stream assembler 485, such as data packets including 512 sample blocks and adaptive noise shaping filter control/synchronization information for the sample blocks.
Fig. 7 is a functional block diagram of an embodiment of an audio signal decoder 700 that may employ Adaptive Differential Pulse Code Modulation (ADPCM). The decoder 700 may be used, for example, as the decoder 430 of fig. 4, as a separate decoder to decode the received encoded signal, and so on. As shown in fig. 7, decoder 700 has a bit stream decomposer 785, an optional codeword decoding circuit 750, an inverse quantizer circuit 734, a predictor circuit 738, an inverse quantizer step size and average bit rate control circuit 740, an adder 770, an adaptive noise shaping inverse filter 780, and a low pass filter 775.
In operation of the embodiment, the assembled signal is received by the bit stream splitter 785 and split into an encoded signal component and an adaptive noise shaping filter control and synchronization signal component. The encoded signal component is provided to a decoding circuit 750, which decoding circuit 750 converts the encoded signal into a quantized signal c n . As discussed above with reference to encoder 450 of fig. 4, escape encoding may be used in embodiments. The quantized signal to be decoded is provided to an inverse quantizer 734 and to an inverse quantizer step size and average bit rate control circuit 740. When decoder 700 is used in an encoder (e.g., encoder 400 of fig. 4), decoding circuit 750 may generally be omitted and the same step size and average bit rate control circuit may be used to provide step size control signals to the step and to the vectorizer and inverse quantizer (see fig. 4).
The inverse quantizer 734 generates a signal such as an analog signal based on the quantized signal output by the decoding circuit 750 (or received from the quantizer (see quantizer 420 of fig. 4)) and the current step size set by the inverse quantizer step size and average bit rate control circuit 740. The output of the inverse quantizer 734 is provided to a first positive input of an adder 770. The output of adder 770 is provided to predictor 738, which predictor 738 is shown to include a Finite Impulse Response (FIR) filter. The output of the FIR filter is provided to a second positive input of adder 770.
When decoder 700 is used as a decoder to provide a decoded signal as an output, the output of decoder 700 is provided to an inverse filter, shown as an adaptive noise-shaping inverse filter 780. The adaptive noise shaping inverse filter 780 may be, for example, a low-order full-polar prediction filter. In one embodiment, the adaptive noise shaping inverse filter 780 is an all-pole adaptive noise shaping filter that recovers the spectrum of the signal using the same coefficients as those used by the corresponding adaptive noise shaping filter of the corresponding encoder (e.g., adaptive noise shaping filter 480 of fig. 4) as the coefficients of the all-pole filter. This information may be conveyed in the bitstream and provided to an adaptive noise shaping inverse filter 780 through a decomposer 785. The setting and synchronization information may be provided periodically, e.g. once every 512 sample blocks. In some embodiments, the adaptive noise shaping inverse filter control information may be, for example, implicit in codewords of the bitstream, as discussed above with reference to fig. 4.
The output of the adaptive noise shaping inverse filter 780 is optionally filtered by a low pass filter 775. This helps remove the recovered high frequency energy when the original spectrum of the signal is recovered by the adaptive noise-shaping inverse filter 780. In one embodiment, the low pass filter 775 of the decoder 700 may employ the same coefficients as those used by the corresponding low pass filter of the encoder (e.g., the low pass filter 475 of fig. 4).
When decoder 700 is used as part of a feedback loop in an encoder (e.g., decoder 430 used in encoder 400 of fig. 4), the output of predictor circuit 738 provides a prediction signal to the encoder (see the prediction signal provided to negative input 414 of adder 410 of fig. 4).
The inverse quantizer 734, the inverse quantizer step size and the average bit rate control circuit 740, and the predictor circuit 738 are generally capable of operating in a similar manner to corresponding components of an encoder (e.g., the encoder 400 of fig. 4). For example, referring to fig. 4 and 7, having the corresponding components operate in a similar manner in encoder 400 and decoder 700 facilitates using the quantized signal to generate a prediction signal and controlling the step size and average bit rate in encoder 400 and decoder 700 without exchanging additional control signals between encoder 400 and decoder 700. For example, a system including an embodiment of encoder 400 and an embodiment of decoder 700 may operate using the same control parameters (e.g., using the same filter coefficients) for the respective components.
As shown, the decoder 700 of fig. 7 includes one or more processors or processor cores P, one or more memories M, and discrete circuits DC, which may be used alone or in various combinations to implement the functions of the decoder 700. Although the components of decoder 700 of fig. 7 are shown as separate components, the various components may be combined (e.g., in some embodiments, inverse quantizer step size and average rate control circuit 740 may be integrated into inverse quantizer 734) or separated into additional components (e.g., predictor circuit 738 may be separated into separate components, such as filters, adders, buffers, look-up tables, etc.), and various combinations thereof.
Fig. 8 is a functional block diagram of an embodiment of a quantizer step size and average rate control circuit 840 that may be used, for example, as the quantizer step size and average bit rate control circuit 440 in the embodiment of the encoder 400 of fig. 4, or as the inverse quantizer step size and average bit rate control circuit 740 in the embodiment of the decoder 700 of fig. 7. As shown, the quantizer step size and average bit rate control circuit 840 includes: multiplier 852 receiving the current quantized signal word c n And a load factor L factor Is the reciprocal of (2); and a logarithmic multiplier selector 842 that selects a logarithmic multiplier based on the current quantized signal word and the load factor. As shown, the current quantized signal word is the word output by the variable rate adaptive quantizer 820. In some embodiments, the current quantized signal word may be included in a bitstream being decoded by a decoder (see fig. 7). The log multiplier selector 842 may select the log multiplier based on historical data (e.g., previously quantized signal words) and may include a look-up table LUT that may be capable of being updated in an update download, e.g., based on the historical data. The logarithmic multiplier selector 842 may select the logarithmic multiplier based on a statistical probability that is based on the current quantized signal word and the previous quantized signal word. The quantization step size and average bit rate control circuit 840 includes an adder 844 that receives the selected logarithmic multiplier at a first positive input and provides an output to a delay circuit 846. The output of delay circuit 846 is provided to multiplier 848 and to exponent circuit 850. Multiplier 848 multiplies the output of delay circuit 846 by a scaling or leakage factor β, which may typically be close to and less than 1, and provides the result to a second positive input of adder 844. The leakage factor may generally be constant, but may be variable in some embodiments, for example, based on a previous step control signal or other historical data. From the following components The errors introduced will decay and so selecting a scale factor β close to and less than 1 helps reduce the impact of selecting an incorrect step size, for example due to transmission errors.
In operation, the exponent circuit 850 generates a step control signal based on the output of the delay circuit 846. As shown, the step size and average bit rate control signal is provided to a variable rate adaptive quantizer 820 and to an inverse quantizer 834. As shown, the quantizer step size and average bit rate control circuit 840 operates in a logarithmic manner, which may simplify the calculation. Some embodiments may operate in a linear fashion and may employ, for example, multipliers instead of adders 844, and exponential circuits instead of multipliers 846, etc. The step size and average bit rate control circuit as shown operates logarithmically and the step size selected based on the step size control signal varies exponentially. In one embodiment, the quantizer step size and average bit rate control circuit 840 may operate according to equation 3 or equation 4 and may select a logarithmic multiplier value to populate the lookup table, as discussed in more detail above with reference to fig. 4 and 6.
As shown, fig. 8 includes one or more processors P, one or more memories M, and discrete circuits DC, which may be used alone or in various combinations to implement the functions of the quantizer step size and average bit rate control circuit 840. The illustrated components, e.g., adders, multipliers, etc., can be implemented in various ways, e.g., using discrete circuitry, executing instructions stored in memory, using look-up tables, etc., as well as various combinations thereof.
Fig. 9 shows an embodiment of a method 900 of generating codewords from an audio signal and controlling the variation of quantizer step size and average bit rate, which method 900 may be employed, for example, by the encoder 400 of fig. 4 when employing escape coding. For convenience, the method 900 will be described with reference to the encoder 400 of fig. 4. The method starts at 902 and proceeds to 904. At 904, the encoder 400 collects a block of audio samples and proceeds to 906. At 906, the encoder 400 processes the samples for each channel. Parallel processing of channel samples may be employed.
At 906a, adaptive quantizer 420 determines whether the channel has audio samples to process. If the channel has audio samples, then method 900 proceeds from 906a to 908. At 908, the encoder 450 determines whether the quantized samples have corresponding symbols in a codebook, which is a huffman codebook as shown. The method proceeds from 908 to 910 when it is determined that the quantized samples have corresponding symbols in the codebook. At 910, the encoder 450 writes the corresponding symbol to the bitstream. Method 900 proceeds from 910 to 914.
When it is determined at 908 that the quantized samples have no corresponding symbols in the codebook, the method 900 proceeds from 908 to 912. At 912, the encoder writes the embedded escape code and quantized sample values into the bitstream, as shown, with the embedded escape code followed by 16-bit quantized sample values. As discussed in more detail above, other methods of transmitting quantized sample values without a corresponding codeword in the codebook may be employed. The method proceeds from 912 to 914.
At 914, the step size and average bit rate control circuit 440 updates the step size control signals for the respective channels, as discussed in more detail above. For example, equation 1, equation 3, and equation 4 may be employed. Method 900 proceeds from 914 to 906 to process the next sample of the channel.
At 906b, the adaptive quantizer determines whether the channel has audio data, but there are no more samples in the block to be processed. For example, the channel may end prematurely. The method 900 proceeds from 906b to 916 when it is determined that the channel has no more samples in the block. At 916, encoder 450 writes the channel end escape code to the bitstream and processing for the channel in the current block is terminated. Method 900 proceeds from 916 to 906.
At 906c, the encoder 400 determines whether all audio data in the blocks of all channels have been processed. When it is determined at 906c that all of the audio data in the block has been processed, the method 900 proceeds from 906c to 918. At 918, the encoder 400 determines if there is more data to start a new block. When it is determined at 918 that there is more data to start a new block, the method 900 proceeds from 918 to 904 where, at 904, a next block of audio samples is processed. When it is determined at 918 that there is no data to begin a new block, the method proceeds to 920. At 920, encoder 450 writes an end-of-stream escape code to the bitstream. The method proceeds from 920 to 930 where processing of the audio signal terminates at 930.
Some embodiments of encoder 400 may perform other acts not shown in fig. 9, may perform less than all of the acts shown in fig. 9, or may perform the acts of fig. 9 in a different order.
Fig. 10 illustrates an embodiment of a method 1000 of generating quantized signal values from codewords, which may be employed by, for example, decoder 700 of fig. 7 when employing escape encoding. Method 1000 may process codewords for multiple channels of a signal in parallel. For convenience, the method 1000 will be described with reference to the decoder 700 of fig. 7. The method starts at 1002 and proceeds to 1004. At 1004, decoding circuit 750 receives a codeword (or a codeword when multiple channels are processed in parallel) and enters 1006.
At 1006, the decoding circuit 750 determines whether the codeword (symbol) has a corresponding quantized sample value in a codebook, such as a huffman codebook. When it is determined that the codeword (symbol) has a corresponding quantized sample value in the codebook, the method 1000 proceeds from 1006 to 1008, where the corresponding quantized sample value is output as the current quantized signal value c by the decoding circuit 750 at 1008 n . Method 1000 proceeds from 1008 to 1004 to process the next codeword of the channel (and codewords of other channels of the encoded signal). When it is determined at 1006 that the codeword (symbol) has no corresponding quantized sample values in the codebook, the method 1000 proceeds from 1006 to 1010.
At 1010, decoding circuit 750 determines whether the codeword is an embedded escape code. When it is determined at 1010 that the codeword is an embedded escape code, method 1000 proceeds from 1010 to 1012, where at 1012 the next codeword of the channel is output by decoding circuit 750 as the current quantized signal value c n . Method 1000 proceeds from 1012 to 1004 to process the next codeword of the channel (and codewords of other channels of the encoded signal). When it is determined at 1010 that the codeword is not an embedded escape code, method 1000 proceeds from 1010 to 1014.
At 1014, decoding circuit 750 determines whether the codeword is the end of the channel escape code. When it is determined at 1014 that the codeword is the end of the channel escape code, method 1000 proceeds from 1014 to 1016, where processing of the signal channel terminates. Method 1000 proceeds from 1016 to 1004 to process the next codeword for the remaining channels of the signal. When it is determined at 1014 that the codeword is not the end of the channel escape code, method 1000 proceeds from 1014 to 1018.
At 1018, decoding circuit 750 determines whether the codeword is the end of the signal escape code. When it is determined at 1018 that the codeword is the end of the signal escape code, the method 1000 proceeds from 1018 to 1020 where processing of the signal terminates at 1020. From 1020, method 1000 proceeds to 1022, where method 1000 terminates at 1022. When it is determined at 1018 that the codeword is not the end of the signal escape code, the method 1000 proceeds from 1018 to 1004 to process the next codeword (or block) of the channel (as well as codewords of other channels of the encoded signal).
Some embodiments of decoder 700 may perform other acts not shown in fig. 10, may perform less than all of the acts shown in fig. 10, or may perform the acts of fig. 10 in a different order.
Some embodiments may take the form of or include a computer program product. For example, according to one embodiment, a computer readable medium is provided, comprising a computer program adapted to perform one or more of the above methods or functions. The medium may be a physical storage medium such as a read-only memory (ROM) chip, or a medium such as a digital versatile disk (DVD-ROM), compact disk (CD-ROM), hard disk, memory, network, or portable media product to be read by an appropriate drive or via an appropriate connection, including other relevant code encoded on one or more bar codes or stored on one or more such computer readable media and readable by an appropriate reading device.
Moreover, in some embodiments, some or all of the methods and/or functions can be implemented or provided in other ways, such as at least partially in firmware and/or hardware, including but not limited to: one or more Application Specific Integrated Circuits (ASICs), digital signal processors, discrete circuits, logic gates, standard integrated circuits, controllers (e.g., by executing appropriate instructions, and including microcontrollers and/or embedded controllers), field Programmable Gate Arrays (FPGAs), complex Programmable Logic Devices (CPLDs), etc., as well as devices employing RFID technology, and various combinations thereof.
The various embodiments described above may be combined to provide further embodiments. Aspects of the embodiments can be modified, if necessary, to employ concepts of the various patents, applications and publications to provide yet other embodiments.
These and other changes can be made to the embodiments in light of the above detailed description. In general, in the following claims, the terms used should not be construed to limit the claims to the specific embodiments disclosed in the specification and the claims, but should be construed to include all possible embodiments along with the full scope of equivalents to which such claims are entitled. Accordingly, the claims are not limited by the present disclosure.

Claims (75)

1. An apparatus for encoding and decoding an audio signal, comprising:
a low pass filter having determined filter coefficients and configured to filter an input signal;
an encoder configured to generate a quantized signal based on a difference signal, and comprising:
an adaptive quantizer; and
a decoder configured to generate a feedback signal and having an inverse quantizer and a predictor circuit having determined control parameters based on a frequency response of the low pass filter, wherein the predictor circuit comprises a finite impulse response, FIR, filter and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter.
2. The apparatus of claim 1, wherein the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter.
3. The apparatus of claim 1, comprising:
an adaptive noise shaping filter coupled between the low pass filter and the encoder, the adaptive noise shaping filter configured to flatten signals within a spectrum corresponding to a spectrum of the low pass filter.
4. The apparatus of claim 3, wherein the adaptive noise shaping filter is configured not to flatten frequencies above an edge frequency of the low pass filter.
5. The apparatus of claim 4, wherein the edge frequency is 25kHz.
6. The apparatus of claim 3, wherein the adaptive noise shaping filter generates a signal indicative of filter coefficients of the adaptive noise shaping filter, the signal indicative of filter coefficients of the adaptive noise shaping filter being included in a bitstream output by the encoder.
7. The apparatus of claim 1, wherein the encoder comprises an encoding circuit configured to generate a codeword based on a quantized signal word generated by the adaptive quantizer.
8. The apparatus of claim 7, wherein the encoding circuit is configured to generate an escape code in response to at least one of:
quantized signal words not associated with a respective encoded codeword;
ending of the signal path of the signal to be encoded; and
and ending the signal to be coded.
9. The apparatus of claim 7, wherein the encoding circuit is configured to generate the codeword using huffman coding.
10. The apparatus of claim 1, wherein the adaptive quantizer is a variable rate quantizer.
11. The apparatus of claim 10, wherein a step size and a bit rate of the quantized signal generated by the adaptive quantizer are variable.
12. The apparatus of claim 10, wherein the adaptive quantizer is configured to control a step size according to the following equation:
dn+1＝βdn+m(cn/L factor )，
where cn is the current quantized signal word, dn corresponds to the current step size in the logarithmic domain, L factor Is the load factor, m (cn/L factor ) Based on the current quantized signal cn and the load factor L factor The selected logarithmic multiplier, β, is the leakage coefficient and dn+1 corresponds to the step size in the logarithmic domain to be applied to the next quantized signal word cn+1.
13. The apparatus of claim 10, wherein the adaptive quantizer is configured to control a step size according to the following equation:
dn+1＝max(βdn+m(cn/Lfactor),dmin)，
where cn is the current quantized signal word, dn corresponds to the current step size in the logarithmic domain, lfactor is the load factor, m (cn/Lfactor) is the logarithmic multiplier selected based on the current quantized signal cn and the load factor Lfactor, β is the leakage factor, dmin is the threshold step size in the logarithmic domain, and dn+1 corresponds to the step size in the logarithmic domain to be applied to the next quantized signal word cn+1.
14. A method for encoding and decoding an audio signal, comprising:
filtering an input signal, the filtering comprising using a low pass filter having determined filter coefficients; and
encoding a filtered input signal using a feedback loop, the encoding comprising:
generating a quantized signal based on the difference signal using an adaptive quantizer;
generating a feedback signal based on the quantized signal using an inverse quantizer and a predictor circuit, the predictor circuit having determined control parameters based on a frequency response of the low pass filter, wherein the predictor circuit comprises a finite impulse response, FIR, filter and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter; and
The difference signal is generated based on the feedback signal and the filtered input signal.
15. The method of claim 14, wherein the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter.
16. The method of claim 15, wherein the filtering comprises filtering the signal output by the low pass filter using an adaptive noise shaping filter that flattens the signal within a spectrum corresponding to a spectrum of the low pass filter.
17. The method of claim 16, comprising:
a signal indicative of the filter coefficients of the adaptive noise shaping filter is generated and included in the encoded bitstream.
18. The method of claim 14, comprising:
a codeword is generated based on the quantized signal word generated by the adaptive quantizer.
19. The method of claim 18, comprising:
generating an escape code in response to at least one of:
quantized signal words not associated with a respective encoded codeword;
Ending of the signal path of the signal to be encoded; and
and ending the signal to be coded.
20. The method of claim 14, comprising:
controlling the step size of the adaptive quantizer according to the following equation:
dn+1＝max(βdn+m(cn/Lfactor),dmin)，
where cn is the current quantized signal word, dn corresponds to the current step size in the logarithmic domain, lfactor is the load factor, m (cn/Lfactor) is the logarithmic multiplier selected based on the current quantized signal cn and the load factor Lfactor, β is the leakage factor, dmin is the threshold step size in the logarithmic domain, and dn+1 corresponds to the step size in the logarithmic domain to be applied to the next quantized signal word cn+1.
21. A non-transitory computer readable medium having contents to configure a signal processing circuit to perform a method comprising:
filtering an input signal, the filtering comprising low pass filtering using a filter coefficient having a determined filter coefficient; and
encoding a filtered input signal using feedback, the encoding comprising:
generating a quantized signal based on the difference signal;
generating a prediction signal based on the quantized signal using determined control parameters based on the low pass filtered frequency response, wherein generating the prediction signal includes using a finite impulse response, FIR, filter, and the determined control parameters include fixed filter coefficients of the FIR filter; and
The difference signal is generated based on the prediction signal and the input signal.
22. The non-transitory computer-readable medium of claim 21, wherein the determined filter coefficients of the low pass filter are fixed filter coefficients of a low pass filter.
23. The non-transitory computer-readable medium of claim 22, wherein the filtering comprises adaptive noise shaping to flatten signals within a spectrum corresponding to a spectrum of the low pass filter.
24. The non-transitory computer-readable medium of claim 21, wherein the method comprises:
controlling a step size of generating the quantized signal according to the following equation:
dn+1＝max(βdn+m(cn/Lfactor),dmin)，
where cn is the current quantized signal word, dn corresponds to the current step size in the logarithmic domain, lfactor is the load factor, m (cn/Lfactor) is the logarithmic multiplier selected based on the current quantized signal cn and the load factor Lfactor, β is the leakage factor, dmin is the threshold step size in the logarithmic domain, and dn+1 corresponds to the step size in the logarithmic domain to be applied to the next quantized signal word cn+1.
25. A system for encoding and decoding an audio signal, comprising:
An encoder, the encoder comprising:
a low pass filter having determined filter coefficients and configured to filter an input signal;
an adaptive quantizer configured to generate a quantized signal based on the difference signal;
an inverse quantizer; and
a predictor circuit coupled between the adaptive quantizer and the predictor circuit, the predictor circuit having a determined control parameter based on a frequency response of the low pass filter, wherein the predictor circuit comprises a finite impulse response, FIR, filter and the determined control parameter of the predictor circuit comprises a fixed filter coefficient of the FIR filter; and
a decoder configured to decode the signal encoded by the encoder.
26. The system of claim 25, wherein the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter.
27. The system of claim 25, comprising:
an adaptive noise shaping filter coupled between the low pass filter and the adaptive quantizer, the adaptive noise shaping filter configured to flatten signals within a spectrum corresponding to a spectrum of the low pass filter.
28. The system of claim 27, wherein the adaptive noise shaping filter generates a signal indicative of filter coefficients of the adaptive noise shaping filter, the signal indicative of filter coefficients of the adaptive noise shaping filter being included in a bitstream output by the encoder to the decoder.
29. The system of claim 25, wherein the encoder comprises an encoding circuit configured to generate codewords based on quantized signal words generated by the adaptive quantizer, the decoder comprising a decoding circuit configured to generate quantized signal words based on codewords generated by the encoding circuit.
30. The system of claim 29, wherein the encoding circuit and the decoding circuit are configured to use escape encoding.
31. A system for encoding and decoding an audio signal, comprising:
an input filter having determined control parameters and configured to limit a bandwidth of an input signal to less than seventy-five percent of an available bandwidth based on a sampling frequency of the input signal;
an encoder configured to generate a quantized signal based on a difference signal, and comprising:
An adaptive quantizer; and
a feedback circuit configured to generate a feedback signal and having an inverse quantizer and a predictor circuit having a determined control parameter based on a frequency response of the input filter, wherein the predictor circuit comprises a finite impulse response, FIR, filter and the determined control parameter of the predictor circuit comprises a fixed filter coefficient of the FIR filter.
32. The system of claim 31, comprising:
a decoder configured to decode the signal encoded by the encoder.
33. The system of claim 31, wherein the input filter is a low pass filter and the determined control parameter of the low pass filter is a fixed filter coefficient of the low pass filter.
34. The system of claim 31, wherein the input filter is a bandpass filter, the determined control parameter of the bandpass filter is a fixed filter coefficient of the bandpass filter, the predictor circuit comprises a finite impulse response FIR filter, and the determined control parameter of the predictor circuit comprises a fixed filter coefficient of the FIR filter.
35. A system for encoding and decoding an audio signal, comprising:
a low-pass filtering means for low-pass filtering the input signal using the determined filtering parameters;
quantized signal generating means for generating a quantized signal based on a difference signal;
a prediction signal generation means that generates a prediction signal based on the quantized signal using a determined control parameter based on a frequency response of the low-pass filter means, wherein the prediction signal generation means includes a finite impulse response FIR filter having fixed filter coefficients based on filter coefficients of the low-pass filter; and
and a difference signal generating means that generates the difference signal.
36. The system of claim 35, comprising:
and a decoding device that decodes the encoded signal.
37. The system of claim 35, wherein the low pass filter means comprises a low pass filter having fixed filter coefficients and the prediction signal generating means comprises a finite impulse response FIR filter having fixed filter coefficients based on the filter coefficients of the low pass filter.
38. An apparatus for encoding and decoding an audio signal, comprising:
a decoder configured to generate a decoded signal based on a quantized signal, the decoder comprising:
an inverse quantizer; and
a predictor circuit; and
a low pass filter having determined filter coefficients and configured to receive an output of the decoder, wherein the predictor circuit has determined control parameters based on a frequency response of the low pass filter, wherein the predictor circuit comprises a finite impulse response, FIR, filter, and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter.
39. The apparatus of claim 38, wherein the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter.
40. The apparatus of claim 38, comprising:
an adaptive noise shaping inverse filter coupled between the inverse quantizer and the low pass filter.
41. An apparatus as defined in claim 40, wherein the adaptive noise-shaping inverse filter is configured to receive a signal included in a bitstream received by the decoder and indicative of adaptive noise-shaping inverse filter coefficients.
42. The apparatus of claim 38, wherein the decoder comprises a decoding circuit configured to generate quantized signal words based on codewords in a bitstream received by the decoder.
43. The apparatus of claim 42, wherein the decoding circuit is configured to respond to at least one of:
an escape code indicating quantized signal words included in the bitstream;
an escape code indicating the end of the signal path; and
an escape code indicating the end of the signal to be encoded.
44. The apparatus of claim 42, wherein the decoding circuitry is configured to decode codewords in the bitstream using huffman coding.
45. The apparatus of claim 38, wherein the inverse quantizer is a variable rate inverse quantizer.
46. The apparatus of claim 38, wherein the inverse quantizer is configured to control step size according to the following equation:
dn+1＝βdn+m(cn/L factor )，
where cn is the current quantized signal word, dn corresponds to the current step size in the logarithmic domain, L factor Is the load factor, m (cn/L factor ) Based on the current quantized signal cn and the load factor L factor The selected logarithmic multiplier, β, is the leakage coefficient and dn+1 corresponds to the step size in the logarithmic domain to be applied to the next quantized signal word cn+1.
47. The apparatus of claim 38, wherein the inverse quantizer is configured to control step size according to the following equation:
dn+1＝max(βdn+m(cn/Lfactor),dmin)，
where cn is the current quantized signal word, dn corresponds to the current step size in the logarithmic domain, lfactor is the load factor, m (cn/Lfactor) is the logarithmic multiplier selected based on the current quantized signal cn and the load factor Lfactor, β is the leakage factor, dmin is the threshold step size in the logarithmic domain, and dn+1 corresponds to the step size in the logarithmic domain to be applied to the next quantized signal word cn+1.
48. A method for encoding and decoding an audio signal, comprising:
decoding an encoded signal using a feedback loop, the decoding comprising:
inverse quantizing the quantized signal using an inverse quantizer; and
generating a prediction signal based on the quantized signal using a predictor circuit; and
filtering the decoded signal using a low pass filter having determined filter coefficients, wherein the predictor circuit has determined control parameters based on a frequency response of the low pass filter, wherein the predictor circuit comprises a finite impulse response, FIR, filter, and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter.
49. A method as defined in claim 48, wherein the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter.
50. A method as defined in claim 49, wherein the filtering comprises using an adaptive noise-shaping inverse filter coupled between an output of a decoder and an input of the low-pass filter.
51. The method of claim 50, comprising:
filter coefficients of the adaptive noise-shaping inverse filter are set based on a signal included in a bitstream of the encoded signal.
52. The method of claim 48, comprising:
quantized signal words are generated based on codewords included in a bitstream of the encoded signal.
53. A method as defined in claim 52, comprising generating the quantized signal word based on the codeword using escape encoding.
54. A method as defined in claim 52, comprising decoding codewords in the bitstream using Huffman coding.
55. A method as defined in claim 48, wherein the inverse quantizer is configured to control the step size according to the following equation:
dn+1＝βdn+m(cn/L factor )，
where cn is the current quantized signal word, dn corresponds to the current step size in the logarithmic domain, L factor Is the load factor, m (cn/L factor ) Based on the current quantized signal cn and the load factor L factor The selected logarithmic multiplier, β, is the leakage coefficient and dn+1 corresponds to the step size in the logarithmic domain to be applied to the next quantized signal word cn+1.
56. A method as defined in claim 48, wherein the inverse quantizer is configured to control the step size according to the following formula:
dn+1＝max(βdn+m(cn/Lfactor),dmin)，
where cn is the current quantized signal word, dn corresponds to the current step size in the logarithmic domain, lfactor is the load factor, m (cn/Lfactor) is the logarithmic multiplier selected based on the current quantized signal cn and the load factor Lfactor, β is the leakage factor, dmin is the threshold step size in the logarithmic domain, and dn+1 corresponds to the step size in the logarithmic domain to be applied to the next quantized signal word cn+1.
57. A non-transitory computer readable medium having contents to configure a signal processing circuit to perform a method comprising:
decoding an encoded signal using feedback, the decoding comprising:
inverse quantizing the quantized signal; and
generating a prediction signal based on the quantized signal; and
Filtering the decoded signal, the filtering comprising a low pass filter using determined filter coefficients, wherein the generating the prediction signal comprises using determined control parameters based on a frequency response of the low pass filter, wherein the prediction signal is generated using a finite impulse response, FIR, filter, and the determined control parameters comprise fixed filter coefficients of the FIR filter.
58. A non-transitory computer readable medium as defined in claim 57, wherein the determined filter coefficients are fixed filter coefficients of a low pass filter.
59. A non-transitory computer readable medium as defined in claim 57, wherein the filtering comprises applying adaptive noise-shaping inverse filtering to the decoded signal.
60. The non-transitory computer readable medium of claim 57, wherein the method comprises:
quantized signal words are generated based on codewords included in a bitstream of the encoded signal.
61. A system for encoding and decoding an audio signal, comprising:
a decoder configured to generate a decoded signal based on a quantized signal, the decoder comprising:
An inverse quantizer; and
a predictor circuit; and
an encoder comprising a low pass filter having determined filter coefficients and configured to filter a signal to be encoded by the encoder, a predictor circuit of the decoder having determined control parameters based on a frequency response of the low pass filter of the encoder, wherein the predictor circuit comprises a finite impulse response, FIR, filter and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter.
62. A system as defined in claim 61, wherein the determined filter coefficients of the low pass filter are fixed filter coefficients of the low pass filter.
63. The system of claim 61, comprising:
an adaptive noise-shaping inverse filter coupled to an output of the inverse quantizer of the decoder.
64. A system as defined in claim 63, wherein the adaptive noise-shaping inverse filter is configured to apply filter coefficients based on a synchronization signal included in a bitstream received by the decoder.
65. A system as defined in claim 61, wherein the decoder comprises decoding circuitry configured to generate quantized signal words based on codewords included in a bitstream received by the decoder from the encoder.
66. A system for encoding and decoding an audio signal, comprising:
a decoder configured to generate a decoded signal based on a quantized signal, the decoder comprising:
an inverse quantizer; and
a predictor circuit; and
an output filter coupled to the decoder and having determined control parameters to limit a bandwidth of an output of the decoder to less than seventy-five percent of an available bandwidth based on a sampling frequency of the quantized signal, wherein the predictor circuit has determined control parameters based on a frequency response of the output filter, wherein the predictor circuit comprises a finite impulse response FIR filter, and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter.
67. The system of claim 66, comprising: an encoder configured to generate an encoded signal.
68. A system as defined in claim 66 in which the output filter is a low pass filter and the determined control parameter of the low pass filter is a fixed filter coefficient of the low pass filter.
69. The system of claim 66 wherein the output filter is a bandpass filter, the determined control parameters of the bandpass filter are fixed filter coefficients of the bandpass filter, the predictor circuit comprises a finite impulse response FIR filter, and the determined control parameters of the predictor circuit comprise fixed filter coefficients of the FIR filter.
70. A system for encoding and decoding an audio signal, comprising:
a decoder configured to generate a decoded signal based on a quantized signal, the decoder comprising:
an inverse quantizer; and
a predictor circuit; and
an output filter configured to filter an output of the decoder, wherein the predictor circuit has a determined control parameter based on a frequency response of an encoder low pass filter, wherein the predictor circuit comprises a finite impulse response, FIR, filter, and the determined control parameter of the predictor circuit comprises a fixed filter coefficient of the FIR filter.
71. A system according to claim 70, comprising an encoder comprising the encoder low pass filter.
72. The system of claim 70, comprising:
an adaptive noise-shaping inverse filter coupled to an output of the inverse quantizer of the decoder.
73. A system for encoding and decoding an audio signal, comprising:
an inverse quantization means that inversely quantizes the quantized signal;
a prediction signal generation means that generates a prediction signal based on the quantized signal, the prediction signal generation means using a determined control parameter based on a frequency response of an encoder low-pass filter, wherein the prediction signal generation means includes a finite impulse response FIR filter, and the finite impulse response FIR filter includes fixed filter coefficients based on filter coefficients of the low-pass filter;
a decoded signal generation means that generates a decoded signal based on the quantized signal and the prediction signal; and
a decoded signal filtering means that filters the decoded signal.
74. A system as defined in claim 73, comprising an encoder including the encoder low pass filter.
75. The system of claim 73, comprising:
spectrum restoration means that restores a spectrum of the decoded signal.
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