WO2017196833A1 - Système de codec audio adaptatif, procédé, appareil et support - Google Patents

Système de codec audio adaptatif, procédé, appareil et support Download PDF

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Publication number
WO2017196833A1
WO2017196833A1 PCT/US2017/031735 US2017031735W WO2017196833A1 WO 2017196833 A1 WO2017196833 A1 WO 2017196833A1 US 2017031735 W US2017031735 W US 2017031735W WO 2017196833 A1 WO2017196833 A1 WO 2017196833A1
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Prior art keywords
signal
step size
filter
quantized signal
quantized
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PCT/US2017/031735
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English (en)
Inventor
James Johnston
Stephen White
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Immersion Services LLC
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Publication date
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 US15/151,109 external-priority patent/US10699725B2/en
Priority claimed from US15/151,211 external-priority patent/US20170330575A1/en
Application filed by Immersion Services LLC filed Critical Immersion Services LLC
Priority to AU2017262757A priority Critical patent/AU2017262757B2/en
Priority to EP17724255.9A priority patent/EP3455854B1/fr
Priority to KR1020187035261A priority patent/KR20190011742A/ko
Priority to CN201780040686.9A priority patent/CN109416913B/zh
Priority to JP2019511820A priority patent/JP7005036B2/ja
Priority to CA3024167A priority patent/CA3024167A1/fr
Publication of WO2017196833A1 publication Critical patent/WO2017196833A1/fr

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Classifications

    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/26Pre-filtering or post-filtering
    • G10L19/265Pre-filtering, e.g. high frequency emphasis prior to encoding
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/02Speech 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
    • G10L19/032Quantisation or dequantisation of spectral components
    • G10L19/035Scalar quantisation
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/26Pre-filtering or post-filtering
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L2019/0001Codebooks

Definitions

  • the description relates to systems, methods and articles to encode and decode audio signals.
  • Differential pulse code modulation may be used to reduce the noise level or the bit rate of an audio signal.
  • a difference between an input audio signal and a predictive signal may be quantized to produce an output encoded data stream of a reduced energy.
  • the predictive signal of an encoder may be generated using a decoder including an inverse quantizer and a prediction circuit.
  • Adaptive differential pulse code modulation varies a size of a quantization step of the quantizer (and inverse quantizer) to increase the efficiency in view of a varying dynamic range of an input signal.
  • 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 including: an adaptive quantizer; and a decoder configured to generate a feedback signal and having an inverse quantizer and a predictor circuit, the predictor circuit having determined control parameters based on a frequency response of the low-pass filter.
  • 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.
  • FIR finite impulse response
  • the apparatus comprises: an adaptive noise shaping filter coupled between the low-pass filter and the encoder, the adaptive noise shaping filter being configured to flatten signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.
  • the adaptive noise shaping filter is configured to not flatten frequencies above an edge frequency of the low-pass filter.
  • the edge frequency is 25 kHz.
  • 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 bit stream output by the encoder.
  • the encoder includes coding circuitry configured to generate code words based on quantized signal words generated by the adaptive quantizer.
  • the coding circuitry is configured to generate an escape code in response to at least one of: a quantized signal word not being associated with a corresponding coding code word; an end of a signal channel of a signal to be encoded; and an end of the signal to be encoded.
  • the coding circuitry is configured to use Huffman coding to generate the code words.
  • the adaptive quantizer is a variable rate quantizer.
  • a step size and bit rate of the quantized signals generated by the adaptive quantizer are variable.
  • the adaptive quantizer is configured to control a step size according to:
  • Cn is a current quantized signal word
  • da corresponds to a current step size in a log domain
  • Lfactor is a loading factor
  • m(cn/Lfactor) is a log multiplier selected based on the current quantized signal cn and the loading factor Lfactor
  • is a leakage coefficient
  • da+i corresponds to a step size in the log domain to be applied to a next quantized signal word Cn+i.
  • the adaptive quantizer is configured to control a step size according to:
  • Cn is a current quantized signal word
  • da corresponds to a current step size in a log domain
  • Lfactor is a loading factor
  • m(cn/Lfactor) is a log multiplier selected based on the current quantized signal cn and the loading factor Lfactor
  • is a leakage coefficient
  • dmin is a threshold step size in the log domain
  • da+i corresponds to a step size in the log domain to be applied to a next quantized signal word Cn+i.
  • a method comprises: filtering an input signal, the filtering including using a low-pass filter having determined filter coefficients; and encoding the filtered input signal using a feedback loop, the encoding including: generating a quantized signal based on a difference signal using an adaptive quantizer; generating a feedback signal based on the quantized signal using an inverse quantizer and a 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.
  • 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.
  • the filtering includes using an adaptive noise shaping filter to filter a signal output by the low-pass filter, the adaptive noise shaping filter flattening signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.
  • the method comprises: generating a signal indicative of filter coefficients of the adaptive noise shaping filter and including the signal indicative of filter coefficients of the adaptive noise shaping filter in an encoded bit stream.
  • the method comprises: generating code words based on quantized signal words generated by the adaptive quantizer.
  • the method comprises: generating an escape code in response to at least one of: a quantized signal word not being associated with a corresponding coding code word; an end of a signal channel of a signal to be encoded; and an end of the signal to be encoded.
  • the method comprises: controlling a step size of the adaptive quantizer according to:
  • Cn is a current quantized signal word
  • da corresponds to a current step size in a log domain
  • Lfactor is a loading factor
  • m(cn/Lfactor) is a log multiplier selected based on the current quantized signal cn and the loading factor Lfactor
  • is a leakage coefficient
  • daaa is a threshold step size in the log domain
  • da+i corresponds to a step size in the log domain to be applied to a next quantized signal word cn+i.
  • a non-transitory computer-readable medium's contents configure signal processing circuitry to perform a method, the method comprising: filtering an input signal, the filtering including low-pass filtering using determined filter coefficients; and encoding the filtered input signal using feedback, the encoding including: generating a quantized signal based on a difference signal; generating a prediction signal based on the quantized signal using determined control parameters based on a frequency response of the low-pass filtering; and generating the difference signal based on the prediction signal and the input signal.
  • the determined filter coefficients of the low-pass filtering are fixed filter coefficients of a low-pass filter
  • the generating the predictor signal comprises using a finite impulse response (FIR) filter
  • the determined control parameters comprise fixed filter coefficients of the FIR filter.
  • the filtering includes adaptive noise shaping to flatten signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.
  • the method comprises: controlling a step size of the generating of the quantized signal according to:
  • Cn is a current quantized signal word
  • da corresponds to a current step size in a log domain
  • Lfactor is a loading factor
  • m(cn/Lfactor) is a log multiplier selected based on the current quantized signal cn and the loading factor Lfactor
  • is a leakage coefficient
  • dmia is a threshold step size in the log domain
  • da+i corresponds to a step size in the log domain to be applied to a next quantized signal word cn+i.
  • a system comprises: an encoder, including: 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 a difference signal; an inverse quantizer; and a predictor circuit, the inverse quantizer being coupled between the adaptive quantizer and the predictor circuit with the predictor circuit having determined control parameters based on a frequency response of the low-pass filter; and a decoder configured to decode signals encoded by the encoder.
  • 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.
  • FIR finite impulse response
  • the system comprises: an adaptive noise shaping filter coupled between the low-pass filter and the adaptive quantizer, the adaptive noise shaping filter being configured to flatten signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.
  • 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 bit stream output by the encoder to the decoder.
  • the encoder includes coding circuitry configured to generate code words based on quantized signal words generated by the adaptive quantizer and the decoder includes decoding circuitry configured to generate quantized signal words based on code words generated by the coding circuitry.
  • the coding circuitry and the decoding circuitry are configured to use escape coding.
  • a system comprises: an input filter having determined control parameters and configured to limit a bandwidth of an input signal to less than seventy-five percent of the available bandwidth based on a sampling frequency of the input signal; an encoder configured to generate quantized signals based on a difference signal and including: an adaptive quantizer; and feedback circuitry configured to generate feedback signals and having an inverse quantizer and a predictor circuit, the predictor circuit having determined control parameters based on a frequency response of the input filter.
  • the system comprises: a decoder configured to decode signals encoded by the encoder.
  • the input filter is a low-pass filter
  • the determined control parameters 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.
  • the input filter is a band-pass filter
  • the determined control parameters of the bandpass filter are fixed filter coefficients of the band-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.
  • a system comprises: means for low-pass filtering an input signal using determined filtering parameters; means for generating a quantized signal based on a difference signal; means for generating a prediction signal based on the quantized signal using determined control parameters based on a frequency response of the means for low- pass filtering; and means for generating the difference signal.
  • the system comprises: means for decoding coded signals.
  • the means for low-pass filtering comprises a low-pass filter having fixed filter coefficients and the means for predicting comprises a finite impulse response (FIR) filter having fixed filter coefficients based on the filter coefficients of the low-pass filter.
  • FIR finite impulse response
  • Figure 1 is a functional block diagram of an embodiment of an ADPCM encoder.
  • Figure 2 is a functional block diagram of an embodiment of an ADPCM decoder.
  • Figure 3 is a functional block diagram of an embodiment of a quantizer step size control circuit.
  • Figure 4 is a functional block diagram of an embodiment of an ADPCM encoder.
  • Figure 5 illustrates an example frequency response of an embodiment of a low pass filter.
  • Figure 6 illustrates an embodiment of a method of controlling changes in adaptive quantizer step sizes.
  • Figure 7 is a functional block diagram of an embodiment of an ADPCM decoder.
  • Figure 8 is a functional block diagram of an embodiment of a quantizer step size and bit rate control circuit.
  • Figure 9 illustrates an embodiment of a method of generating code words and controlling changes in adaptive quantizer step sizes.
  • Figure 10 illustrates an embodiment of a method of generating a quantized signal value from a code word.
  • FIG. 1 is a functional block diagram of an embodiment of audio signal encoder 100 which may employ adaptive differential pulse-code modulation (ADPCM). As illustrated in Figure 1 , the encoder 100 has an adder circuit 1 10, an adaptive quantizer circuit 120, a decoder circuit 130 including an inverse quantizer circuit 134 and a predictor circuit 138, a quantizer step size control circuit 140, and an optional coder circuit 150.
  • ADPCM adaptive differential pulse-code modulation
  • an analog input audio signal to be encoded is received at a positive input 112 of the adder 110 of the encoder 100.
  • a negative input 1 14 of the adder 110 receives a prediction signal generated by the decoder 130 as a feedback signal.
  • the adder 110 generates a difference signal which is provided to the adaptive quantizer circuit 120.
  • the adaptive quantizer circuit 120 may be an analog to digital converter which samples the received difference signal and generates an output signal representing the difference signal as a series of quantized signals representing different signal levels. For example, 8-bit words may be used to represent 256 different signal levels (e.g. , 256 different steps having a uniform step size); 4 bits words may be used to represent 16 different signal levels; etc.
  • coding such as Huffman coding and/or arithmetic coding
  • coding circuit 150 may be employed on the quantized signal in an embodiment, by coding circuit 150, generating a coded signal output.
  • the quantized signal output by the adaptive quantizer circuit 120 (or of the optional coder 150 when a coder is employed) is the output quantized signal or code words of the encoder 100.
  • the quantizer step size control circuit 140 generates control signals to control a size of the quantization steps employed by the quantizer 120 (and the inverse quantizer 134), which may be varied to facilitate efficient transmission, storage, etc., in view of an input audio signal having a varying dynamic range.
  • 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 the prediction signal based on the output signal of the inverse quantizer 134 and historical data, such as recent quantized signal values and recent prediction signal values.
  • One or more filters and one or more feedback loops may be employed by the predictor circuit 138.
  • the encoder 100 of Figure 1 comprises one or more processors or processor cores P, one or more memories M, and discrete circuitry DC, which may be used alone or in various combinations to implement the functionality of the encoder 100.
  • an embodiment of the encoder 100 generates quantized and, optionally, coded data from an input analog audio signal.
  • a digital audio signal to be encoded e.g., to a reduced bitstream, may be received at the positive input 112 instead of an analog signal (e.g. , an 8-bit digital audio signal may be encoded as a 4-bit digital audio signal).
  • the various components may be combined (e.g. , the quantizer step size control circuit 140 may be integrated into the adaptive quantizer 120 in some embodiments) or split into additional components (e.g. , the predictor circuit 138 may be split into multiple predictor circuits, may be split into separate 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 which may employ adaptive differential pulse-code modulation (ADPCM).
  • the decoder 200 may be employed, for example, as the decoder 130 of Figure 1, as a separate decoder to decode a received encoded signal, etc.
  • the decoder 200 has optional decoding circuitry 250, an inverse quantizer circuit 234, a predictor circuit 238, an inverse quantizer step size control circuit 240 and an adder 270.
  • a coded signal is received by the decoding circuitry 250, which converts the coded signal into a quantized signal.
  • the quantized signal to be decoded is provided to the inverse quantizer 234 and to the inverse quantizer step size control circuit 240.
  • the decoding circuitry 250 may typically be omitted and the same step size control circuit may be used to provide a step size control signal to the quantizer and to the inverse quantizer (see, Figure 1).
  • the inverse quantizer 234 generates a signal, such as an analog signal, based on the quantized signal output by the decoding circuitry 250 (or received from a quantizer (see quantizer 120 of Figure 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 the adder 270.
  • the output of the adder is provided to the predictor 238, which as illustrated comprises a Finite Impulse Response (FIR) filter.
  • An output of the FIR filter is provided to a second positive input of the adder 270.
  • FIR Finite Impulse Response
  • the output of the decoder 200 is the output of the adder 270.
  • the output of the predictor circuit 238 provides the prediction signal to the encoder (see the prediction signal provided to the negative input 114 of the adder 110 of Figure 1).
  • the inverse quantizer 234, the inverse quantizer step size control circuit 240 and the predictor circuit 238 may typically operate in a similar manner to the corresponding components of an encoder, such as the encoder 100 of Figure 1.
  • an encoder such as the encoder 100 of Figure 1.
  • having the corresponding components operate in a similar manner in the encoder 100 and the decoder 200 facilitates using the quantized signal to generate the prediction signal and to control the step size in both the encoder 100 and the decoder 200, without needing to exchange additional control signals between the encoder 100 and the decoder 200.
  • the decoder 200 of Figure 2 comprises one or more processors or processor cores P, one or more memories M, and discrete circuitry DC, which may be used alone or in various combinations to implement the functionality of the decoder 200.
  • the components of the decoder 200 of Figure 2 are illustrated as separate components, the various components may be combined (e.g. , the inverse quantizer step size control circuit 240 may be integrated into the inverse quantizer 234 in some embodiments) or split into additional components (e.g. , the predictor circuit 238 may be split 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, which may be employed, for example, in the embodiment of the encoder 100 of Figure 1 as the quantizer step size control circuit 140, or in the embodiment of the decoder 200 of Figure 2 as the inverse quantizer step size control circuit 240.
  • the quantizer step size control circuit 340 comprises a log multiplier selector 342 which selects a log multiplier based on a current quantized signal word, as illustrated a word output by an adaptive quantizer 320.
  • the current quantized signal word may be included in a bit stream being decoded by a decoder (see Figure 2).
  • the log multiplier selector 342 may select a log multiplier based on historical data, such as previous quantized signal words, and may comprise a look-up table LUT, which may be updatable, for example, based on historical data, in a update download, etc.
  • the log multiplier selector 342 may select a log multiplier based on statistical probabilities based on current and previous quantized signal words.
  • the quantizer step size control circuit 340 comprises an adder 344 which receives at a first positive input the selected log multiplier, and provides an output to a delay circuit 346. The output of the delay circuit 346 is provided to a multiplier 348 and to an exponential circuit 350.
  • the multiplier 348 multiplies the output of the delay circuit 346 by a scaling or leakage factor ⁇ , which may typically be close to and less than 1, and provides the result to a second positive input of the adder 344.
  • the leakage factor may typically be a constant, but may be variable in some embodiments, for example, based on the previous step size control signal or other historical data.
  • the selection of a scaling factor ⁇ as close to and less than 1 facilitates reducing the impact of selection of an incorrect step size, for example due to a transmission error, as the introduced error will decay away.
  • the exponential circuit 350 in operation, generates a step-size control signal based on the output of the delay circuit 346.
  • the step-size control signal is provided to the adaptive quantizer 320 and to an inverse quantizer 334.
  • the quantizer step size control circuit 340 operates in a logarithmic manner, which may simplify the calculations. Some embodiments may operate in a linear manner, and may, for example, employ a multiplier instead of the adder 244, and an exponential circuit instead of the multiplier 246.
  • the quantizer step-size control circuit 340 as illustrated operates in a logarithmic manner, and the step sizes selected based on the step size control signal vary in an exponential manner.
  • the quantizer step size control circuit 340 may operate in accordance with equation 1, below:
  • Figure 3 comprises one or more processors P, one or more memories M, and discrete circuitry DC, which may be used alone or in various combinations to implement the functionality of the quantizer step size control circuit 340.
  • FIG 4 is a functional block diagram of an audio signal encoder 400 which may employ adaptive differential pulse-code modulation (ADPCM).
  • the audio signal encoder 400 of an embodiment provides added bandwidth control, facilitates avoiding quantizer overload, and includes adaptive noise shaping.
  • the 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, a coder 450 and bit stream assembler 485.
  • ADPCM adaptive differential pulse-code modulation
  • an analog input audio signal to be encoded is received at an input of an input filter, as illustrated the low pass filter 475.
  • the low pass filter 475 facilitates improving the signal to noise ratio.
  • the low pass filter 475 may, for example, be a FIR filter having a 25 kHz edge and a 30 kHz stop band, which has been found to provide excellent results for data sampled at 88.2 or 96 kHz.
  • Figure 5 illustrates an example frequency response of an embodiment of the low pass filter 475 applied to a sampling rate of 96 kHz.
  • the predictor employing filter coefficients based on the frequency response of the input filter facilitates obtaining a substantial prediction gain for an input signal when a sufficiently high sampling rate is employed, which in turn facilitates obtaining a desired minimum signal to noise ratio.
  • sampling rates below 48 kHz e.g. , 44.1 and 48 kHz
  • the output of the low pass filter 475 is provided to the adaptive noise shaping filter 480.
  • 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 to the low pass filter 475.
  • the adaptive noise shaping filter 480 may be omitted or selectively bypassed.
  • the adaptive noise shaping filter 480 may be omitted or bypassed when high bit rate signal encoding is employed.
  • a band pass filter may be employed instead of a low pass filter, with correspond adjustments to the predictor filter.
  • an input filter e.g.
  • a band pass filter having fixed control parameters and configured to limit a bandwidth of an input signal to less than seventy-five percent of the available bandwidth based on the sampling frequency
  • the corresponding decoder may include a predictor circuit having fixed control parameters based on a frequency response of the filter. Limiting the bandwidth of the input signal using the input filter and setting the control parameters of the predictor circuit based on a frequency response of the input filter facilitates obtaining a substantial prediction gain for an input signal when a sufficiently high sampling rate is employed, which in turn facilitates obtaining a desired minimum signal to noise ratio.
  • the adaptive noise shaping filter 480 may be, for example, a low-order all-zero linear prediction filter. Real (not complex) coefficients may be employed.
  • the adaptive noise shaping filter 480 is an all zero adaptive noise shaping filter which flattens the spectrum of the signal received from the low pass filter 475, while maintaining the overall spectral slope and sufficient masking to maintain a transparent codec (e.g., the compression artifacts are generally imperceptible).
  • a transparent codec e.g., the compression artifacts are generally imperceptible
  • an all-pole filter using the same coefficients may be used to restore the original spectral shape.
  • the adaptive noise shaping filter 480 preserves the whiteness criteria for the predictor circuit 438.
  • the low-order noise shaping filter 480 may be adjusted to not flatten signals over an edge frequency of a low-pass filter (e.g. 25 kHz, which may not exist in a signal filtered by a low pass filter 475). As noted above, the missing energy at high frequencies facilitates a higher prediction gain. Filters other than linear prediction filters may be employed as the noise shaping filters.
  • the adaptive noise shaping filter 480 provides a filtered output signal to a positive input 412 of the adder 410.
  • the adaptive noise shaping filter 480 also provides a signal including adaptive noise filter setting information and/or synchronization information, which may be used to communicate adaptive noise filter setting and synchronization information to a decoder, such as the decoder 700 of Figure 7, which includes a corresponding inverse noise shaping filter 780.
  • the setting and synchronization information may be transmitted periodically, such as once for every 512 sample block.
  • the adaptive noise shaping filter control information may be implicit in the code words of the bit stream. For example, when the code words of the bit stream indicate an average bit rate above a threshold average bit rate is being employed, this may also indicate that adaptive noise shaping is being bypassed.
  • a negative input 414 of the adder 410 receives a prediction signal generated by the decoder 430 as a feedback signal.
  • the adder 410 generates a difference signal which is provided to the 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 quantization signals or words.
  • the size of the quantization signals is not fixed, and the average length may be adjusted using the output of a multiplier table of a step size and average bit rate controller 440, as discussed in more detail below.
  • the output of the variable rate adaptive quantizer circuit 420 is provided to the step size and average bit rate controller 440, the inverse quantizer 434 and the coder 450.
  • the quantizer step and average bit rate control circuit 440 generates one or more control signals to control a size of the quantization steps. This implicitly determines an average length of the quantization signal employed by the quantizer 420 (and the inverse quantizer 434), which may be varied by adjustment of the multiplier table to facilitate efficient coding in view of an input audio signal having a varying dynamic range.
  • Figure 6 illustrates an embodiment of a method 600 of generating code words and controlling changes in step sizes and average bit rate that may be employed, for example, by the encoder 400 of Figure 4.
  • the method 600 will be described with reference to the encoder 400 of Figure 4.
  • the method starts at 602 and proceeds to 604.
  • the variable rate adaptive quantizer 420 generates a current quantization signal or word based on the difference signal and the current quantization step size control signal. This may be done, for example, in accordance with equation 2, below:
  • Cn is the current quantized signal
  • en is the error or difference signal
  • da corresponds to the current step size in the log domain.
  • the quantizer step size and average bit rate control circuit 440 generates one or more control signals to set the step size for the next quantization signal word. This may be done, for example, in accordance with equation 1, above, or in accordance with equation 3 or 4, below:
  • a minimum step size c min in the log domain may be set, as follows:
  • the loading factor Lfactor may be selected so as 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 log multiplier m(cn/Lfactor) changes the bit rate and step size, and the values stored in the look-up-table of the log multiplier selector (see Figure 8) may be selected so as to cause the adaptive quantizer 420 and inverse quantizer 434 to implement the desired changes in the step size and bit rate. For example, higher log multipliers may indicate an increased step size and lower bit rate to the quantizer 420 and inverse quantizer 434.
  • the look-up table may be indexed based on the result of the current quantization value cn divided by the loading factor Lfactor.
  • values in a look-up-table may be selected such that the log multiplier monotonically increases as the current quantization value cn increases, and the table of multipliers may go from a negative value for small cnto a positive value for large cn.
  • the method 600 proceeds from 606 to 608.
  • the encoder 400 determines whether to continue encoding of a received signal. When it is determined at 608 to continue encoding of a received signal, the method returns to 604 to process the next quantized signal word. When it is not determined at 608 to continue encoding of a received signal, the method proceeds to 610, where other processing may occur, such as generating an escape code to indicate the received signal has terminated, etc. The method proceeds from 610 to 612, where the method 600 terminates.
  • an encoder 400 may perform other acts not shown in Figure 6, may not perform all of the acts shown in Figure 6, or may perform the acts of Figure 6 in a different order.
  • the inverse quantizer 434 of the decoder 430 generates a signal, such as an analog signal, based on the quantized signal output cn by the variable rate adaptive quantizer 420 and the current step size da.
  • the predictor circuit 438 may generate the prediction signal based on the output signal of the inverse quantizer 434 and historical data, such as recent coded data and recent prediction values, as discussed in more detail below with reference to Figure 7.
  • the predictor circuit 438 may employ a FIR filter with coefficients selected based on the frequency response of the low-pass filter 475, as discussed in more detail below with reference to Figure 7. These coefficients may be fixed, and may be selected so as to facilitate maintaining a sufficient signal to noise ratio for anticipated input signal characteristics.
  • the quantized signal output by the variable rate adaptive quantizer circuit 420 (or of the optional coder 450 when a coder is employed) is the output quantized signal of the encoder 400.
  • coding such as Huffman coding and/or arithmetic coding, may be employed on the quantized signal in an embodiment, by coding circuit 450, generating a coded signal output of the encoder 400.
  • the coder 450 converts quantized signal words into code words, for example, using one or more look-up tables. Quantized signal words which are used less frequently may be assigned to larger code words, and quantized signal words which are used more frequently may be assigned to smaller code words to increase the efficiency of the coder 400.
  • the coder 450 optionally provides escape coding in an embodiment.
  • escape code may be sent instead of a code word from the code book, with the escape coding indicating how the quantized signal value or information will be transmitted (e.g. , that the actual quantized signal is being transmitted, that the next code word is the quantized signal value instead of a code word, that a difference between a maximum/minimum level is being transmitted, etc.).
  • an escape code may indicate that a channel of an encoded signal is being discontinued or is not present (e.g. , only one channel of a stereo signal is being encoded).
  • an escape code may indicate an end of an encoded signal.
  • the bit stream assembler 485 receives the code words output by the coder 450 and the adaptive noise shaping filter control/synchronization information output by the adaptive noise shaping filter 480 and assembles a bit stream for transmission to a decoder and/or storage.
  • data packets may be assembled by the bit stream assembler 485, such as packets including a 512 sample block and adaptive noise shaping filter control/synchroniza- tion information for the sample block.
  • FIG. 7 is a functional block diagram of an embodiment of an audio signal decoder 700 which may employ adaptive differential pulse-code modulation (ADPCM).
  • the decoder 700 may be employed, for example, as the decoder 430 of Figure 4, as a separate decoder to decode a received encoded signal, etc.
  • the decoder 700 has a bit stream disassembler 785, optional code word decoding circuitry 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 inverse adaptive noise shaping filter 780 and a low pass filter 775.
  • ADPCM adaptive differential pulse-code modulation
  • an assembled signal is received by the bit stream disassembler 785 and split into a coded signal component and an adaptive noise shaping filter control and synchronization signal component.
  • the coded signal component is provided to the decoding circuitry 750, which converts the coded signal into a quantized signal Cn. Escape coding may be used in an embodiment, as discussed above with reference to the coder 450 of Figure 4.
  • the quantized signal to be decoded is provided to the inverse quantizer 734 and to the inverse quantizer step size and average bit rate control circuit 740.
  • the decoding circuitry 750 may typically be omitted and the same step size and average bit rate control circuit may be used to provide a step size control signal to the quantizer and to the inverse quantizer (see, Figure 4).
  • the inverse quantizer 734 generates a signal, such as an analog signal, based on the quantized signal output by the decoding circuitry 750 (or received from a quantizer (see quantizer 420 of Figure 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 the adder 770.
  • the output of the adder 770 is provided to the predictor 738, which as illustrated comprises a Finite Impulse Response (FIR) filter. An output of the FIR filter is provided to a second positive input of the adder 770.
  • FIR Finite Impulse Response
  • the output of the decoder 700 is provided to an inverse filter, as illustrated an inverse adaptive noise shaping filter 780.
  • the inverse adaptive noise shaping filter 780 may be, for example, a low-order all pole linear prediction filter.
  • the inverse adaptive noise shaping filter 780 is an all-pole adaptive noise shaping filter which restores the spectrum of the signal using the using the same coefficients used by a corresponding adaptive noise shaping filter of a corresponding encoder (e.g. , the adaptive noise shaping filter 480 of Figure 4) as the coefficients of the all-pole filter.
  • This information may be conveyed in the bitstream and provided to the inverse adaptive noise shaping filter 780 by the disassembler 785.
  • the setting and synchronization information may be provided periodically, such as once for every 512 sample block.
  • the inverse adaptive noise shaping filter control information may be implicit in the code words of the bit stream, for example, as discussed above with reference to Figure 4.
  • the output of the inverse adaptive noise shaping filter 780 is optionally filtered by a low-pass filter 775. This facilitates removing high-frequency energy restored when the original spectrum of the signal is restored by the inverse adaptive noise shaping filter 780.
  • the low-pass filter 775 of the decoder 700 may employ the same coefficients used by a corresponding low-pass filter of an encoder (e.g. , the low-pass filter 475 of Figure 4).
  • the output of the predictor circuit 738 provides the prediction signal to the encoder (see the prediction signal provided to the negative input 414 of the adder 410 of Figure 4).
  • the inverse quantizer 734, the inverse quantizer step and average bit rate control circuit 740 and the predictor circuit 738 may typically operate in a similar manner to the corresponding components of an encoder, such as the encoder 400 of Figure 4.
  • an encoder such as the encoder 400 of Figure 4.
  • having the corresponding components operate in a similar manner in the encoder 400 and the decoder 700 facilitates using the quantized signal to generate the prediction signal and to control the step size and average bit rate in both the encoder 400 and the decoder 700, without needing to exchange additional control signals between the encoder 400 and the decoder 700.
  • a system including an embodiment of the encoder 400 and an embodiment of the decoder 700 may operate using the same control parameters for the corresponding components (e.g., using the same filter coefficients).
  • the decoder 700 of Figure 7 comprises one or more processors or processor cores P, one or more memories M, and discrete circuitry DC, which may be used alone or in various combinations to implement the functionality of the decoder 700.
  • the components of the decoder 700 of Figure 7 are illustrated as separate components, the various components may be combined (e.g. , the inverse quantizer step and average rate control circuit 740 may be integrated into the inverse quantizer 734 in some embodiments) or split into additional components (e.g. , the predictor circuit 738 may be split 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, which may be employed, for example, in the embodiment of the encoder 400 of Figure 4 as the quantizer step size and average bit rate control circuit 440, or in the embodiment of the decoder 700 of Figure 7 as the inverse quantizer step size and average bit rate control circuit 740.
  • the quantizer step size and average bit rate control circuit 840 comprises a multiplier 852, which receives a current quantized signal word Cn and an inverse of a loading factor Lfactor, and a log multiplier selector 842 which selects a log multiplier based on the current quantized signal word and the loading factor.
  • the current quantized signal word is a word output by variable rate adaptive quantizer 820.
  • the current quantized signal word may be included in a bit stream being decoded by a decoder (see Figure 7).
  • the log multiplier selector 842 may select a log multiplier based on historical data, such as previous quantized signal words, and may comprise a look-up table LUT, which may be updatable, for example, based on historical data, in a update download, etc.
  • the log multiplier selector 842 may select a log multiplier based on statistical probabilities based on current and previous quantized signal words.
  • the quantized step size and average bit rate control circuit 840 comprises an adder 844 which receives at a first positive input the selected log multiplier, and provides an output to a delay circuit 846.
  • the output of the delay circuit 846 is provided to a multiplier 848 and to an exponential circuit 850.
  • the multiplier 848 multiplies the output of the delay circuit 846 by a scaling or leakage factor ⁇ , which may typically be close to and less than 1 , and provides the result to a second positive input of the adder 844.
  • the leakage factor may typically be a constant, but may be variable in some embodiments, for example, based on the previous step size control signal or other historical data.
  • the selection of a scaling factor ⁇ as close to and less than 1 facilitates reducing the impact of selection of an incorrect step size, for example due to a transmission error, as the introduced error will decay away.
  • the exponential circuit 850 in operation, generates a step-size control signal based on the output of the delay circuit 846.
  • the step-size and average bit rate control signal is provided to a variable rate adaptive quantizer 820 and to an inverse quantizer 834.
  • the quantizer step size and average bit rate control circuit 840 operates in a logarithmic manner, which may simplify the calculations. Some embodiments may operate in a linear manner, and may, for example, employ a multiplier instead of the adder 844, and an exponential circuit instead of the multiplier 846, etc.
  • the step-size and average bit rate control circuit as illustrated operates in a logarithmic manner, and the step sizes selected based on the step size control signal vary in an exponential manner.
  • the quantizer step size and average bit rate control circuit 840 may operate in accordance with equations 3 or equation 4, and select log multiplier values to populate the look-up tables as discussed above in more detail with reference to Figures 4 and 6.
  • Figure 8 comprises one or more processors P, one or more memories M, and discrete circuitry DC, which may be used alone or in various combinations to implement the functionality of the quantizer step size and average bit rate control circuit 840.
  • the illustrated components such as adders, multiplier, etc., may be implemented in various ways, such as, using discrete circuitry, executing instructions stored in a memory, using lookup tables, etc., and various combinations thereof.
  • Figure 9 illustrates an embodiment of a method 900 of generating code words from an audio signal and controlling changes in quantizer step sizes and average bit rate that may be employed, for example, by the encoder 400 of Figure 4 when escape coding is employed.
  • the method 900 will be described with reference to the encoder 400 of Figure 4.
  • the method starts at 902 and proceeds to 904.
  • the encoder 400 collects a block of audio samples and proceeds to 906.
  • the encoder 400 processes a sample of each channel. Parallel processing of the samples of the channels may be employed.
  • the adaptive quantizer 420 determines whether the channel has an audio sample to be processed. If the channel has an audio sample, the method 900 proceeds from 906a to 908.
  • the coder 450 determines whether a quantized sample has a corresponding symbol in a code book, as illustrated, a Huffman code book. When it is determined that the quantized sample has a corresponding symbol in the code book, the method proceeds from 908 to 910.
  • the coder 450 writes the corresponding symbol into the bitstream. The method 900 proceeds from 910 to 914.
  • the method 900 proceeds from 908 to 912.
  • the coder writes an embed escape code and a quantized sample value into the bitstream, as illustrated an embed escape code followed by a 16 bit quantized sample value.
  • Other methods of transmitting a quantized sample value without a corresponding code word in the code book may be employed, as discussed in more detail above. The method proceeds from 912 to 914.
  • the step-size and average bit rate control circuit 440 updates the step size control signal for the corresponding channel, as discussed in more detail above. For example, the equations 1, 3 and 4 may be employed.
  • the method 900 proceeds from 914 to 906 to process the next sample for the channel.
  • the adaptive quantizer determines whether the channel had audio data, but has no more samples in the block to be processed. For example, a channel may have ended prematurely.
  • the method 900 proceeds from 906b to 916.
  • the coder 450 writes an end-of-channel escape code into the bitstream and processing of the channel in the current block terminates. The method 900 proceeds from 916 to 906.
  • the encoder 400 determines whether all the audio data in the block for all of the channels has been processed. When it is determined at 906c that all the audio data in the block has been processed, the method 900 proceeds from 906c to 918. At 918, the encoder 400 determines whether 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 the next block of audio samples is processed. When it is not determined at 918 that there is data to start a new block, the method proceeds to 920. At 920, the coder 450 writes an end of stream escape code into the bit stream. The method proceeds from 920 to 930, where processing of the audio signal terminates.
  • an encoder 400 may perform other acts not shown in Figure 9, may not perform all of the acts shown in Figure 9, or may perform the acts of Figure 9 in a different order.
  • Figure 10 illustrates an embodiment of a method 1000 of generating a quantized signal value from a code word that may be employed, for example, by the decoder 700 of Figure 7 when escape coding is employed.
  • the method 1000 may process code words for multiple channels of a signal in parallel. For convenience, the method 1000 will be described with reference to the decoder 700 of Figure 7.
  • the method starts at 1002 and proceeds to 1004.
  • the decoding circuitry 750 receives a code word (or code words when multiple channels are being processed in parallel) and proceeds to 1006.
  • the decoding circuitry 750 determines whether the code word (symbol) has a corresponding quantized sample value in a code book, such as a Huffman code book. When it is determined that the code word (symbol) has a corresponding quantized sample value in a code book, the method 1000 proceeds from 1006 to 1008, where the corresponding quantized sample value is output by the decoding circuitry 750 as the current quantized signal value Cn. The method 1000 proceeds from 1008 to 1004 to process the next code word of the channel (and code words of other channels of the coded signal). When it is not determined at 1006 that the code word (symbol) has a corresponding quantized sample value in a code book, the method 1000 proceeds from 1006 to 1010.
  • the decoding circuitry 750 determines whether the code word is an embed escape code. When it is determined at 1010 that the code word is an embed escape code, the method 1000 proceeds from 1010 to 1012, where the next code word of the channel is output by the decoding circuitry 750 as the current quantized signal value Cn. The method 1000 proceeds from 1012 to 1004 to process the next code word of the channel (and code words of other channels of the coded signal). When it is not determined at 1010 that the code word is an embed escape code, the method 1000 proceeds from 1010 to 1014.
  • the decoding circuitry 750 determines whether the code word is an end of channel escape code. When it is determined at 1014 that the code word is an end of channel escape code, the method 1000 proceeds from 1014 to 1016, where processing of the signal channel is terminated. The method 1000 proceeds from 1016 to 1004 to process the next code word of the remaining channels of the signal. When it is not determined at 1014 that the code word is an end of channel escape code, the method 1000 proceeds from 1014 to 1018.
  • the decoding circuitry 750 determines whether the code word is an end of signal escape code. When it is determined at 1018 that the code word is an end of signal escape code, the method 1000 proceeds from 1018 to 1020, where processing of the signal is terminated. The method 1000 proceeds from 1020 to 1022 where the method 1000 terminates. When it is not determined at 1018 that the code word is an end of signal escape code, the method 1000 proceeds from 1018 to 1004 to process the next code word (or block) of the channel (and code words of other channels of the coded signal).
  • Some embodiments of a decoder 700 may perform other acts not shown in Figure 10, may not perform all of the acts shown in Figure 10, or may perform the acts of Figure 10 in a different order.
  • a computer readable medium comprising a computer program adapted to perform one or more of the methods or functions described above.
  • the medium may be a physical storage medium, such as for example a Read Only Memory (ROM) chip, or a disk such as a Digital Versatile Disk (DVD-ROM), Compact Disk (CD-ROM), a hard disk, a memory, a network, or a portable media article to be read by an appropriate drive or via an appropriate connection, including as encoded in one or more barcodes or other related codes stored on one or more such computer-readable mediums and being readable by an appropriate reader device.
  • ROM Read Only Memory
  • DVD-ROM Digital Versatile Disk
  • CD-ROM Compact Disk
  • some or all of the methods and/or functionality may be implemented or provided in other manners, 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 circuitry, 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 that employ RFID technology, and various combinations thereof.
  • ASICs application-specific integrated circuits
  • DSPs digital signal processors
  • discrete circuitry discrete circuitry
  • logic gates e.g. , logic gates, standard integrated circuits
  • controllers e.g. , by executing appropriate instructions, and including microcontrollers and/or embedded controllers
  • FPGAs field-programmable gate arrays
  • CPLDs complex programmable logic devices

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Abstract

Un codeur comprend un filtre passe-bas pour filtrer des signaux audio d'entrée. Les coefficients de filtre du filtre passe-bas sont fixes. Le codeur génère des signaux quantifiés sur la base d'un signal de différence. Le codeur comprend un quantificateur adaptatif et un décodeur pour générer des signaux de rétroaction. Le décodeur comprend un quantificateur inverse et un prédicteur. Le prédicteur a des paramètres de commande fixes qui sont basés sur une réponse en fréquence du filtre passe-bas. Le prédicteur peut comprendre un filtre à réponse impulsionnelle finie dont les coefficients de filtre sont fixes. Le décodeur peut comprendre un filtre adaptatif de mise en forme du bruit couplé entre le filtre passe-bas et le codeur. Le filtre adaptatif de mise en forme du bruit permet d'aplanir les signaux dans un spectre de fréquence correspondant à un spectre de fréquence du filtre passe-bas.
PCT/US2017/031735 2016-05-10 2017-05-09 Système de codec audio adaptatif, procédé, appareil et support WO2017196833A1 (fr)

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AU2017262757A AU2017262757B2 (en) 2016-05-10 2017-05-09 Adaptive audio codec system, method, apparatus and medium
EP17724255.9A EP3455854B1 (fr) 2016-05-10 2017-05-09 Procédé de codec audio adaptatif et appareil
KR1020187035261A KR20190011742A (ko) 2016-05-10 2017-05-09 적응형 오디오 코덱 시스템, 방법, 장치 및 매체
CN201780040686.9A CN109416913B (zh) 2016-05-10 2017-05-09 自适应音频编解码系统、方法、装置及介质
JP2019511820A JP7005036B2 (ja) 2016-05-10 2017-05-09 適応オーディオコーデックシステム、方法および媒体
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US15/151,200 US10770088B2 (en) 2016-05-10 2016-05-10 Adaptive audio decoder system, method and article
US15/151,109 2016-05-10
US15/151,109 US10699725B2 (en) 2016-05-10 2016-05-10 Adaptive audio encoder system, method and article
US15/151,211 US20170330575A1 (en) 2016-05-10 2016-05-10 Adaptive audio codec system, method and article
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Citations (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5974380A (en) * 1995-12-01 1999-10-26 Digital Theater Systems, Inc. Multi-channel audio decoder
US6493664B1 (en) * 1999-04-05 2002-12-10 Hughes Electronics Corporation Spectral magnitude modeling and quantization in a frequency domain interpolative speech codec system
US6975254B1 (en) * 1998-12-28 2005-12-13 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Methods and devices for coding or decoding an audio signal or bit stream
US20070118362A1 (en) * 2003-12-15 2007-05-24 Hiroaki Kondo Audio compression/decompression device
US20090254783A1 (en) * 2006-05-12 2009-10-08 Jens Hirschfeld Information Signal Encoding
US20110224995A1 (en) * 2008-11-18 2011-09-15 France Telecom Coding with noise shaping in a hierarchical coder
US20140229186A1 (en) * 2002-09-04 2014-08-14 Microsoft Corporation Entropy encoding and decoding using direct level and run-length/level context-adaptive arithmetic coding/decoding modes

Family Cites Families (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6751587B2 (en) * 2002-01-04 2004-06-15 Broadcom Corporation Efficient excitation quantization in noise feedback coding with general noise shaping
JP3748261B2 (ja) * 2003-06-17 2006-02-22 沖電気工業株式会社 Adpcm方式復号器
JP5129115B2 (ja) * 2005-04-01 2013-01-23 クゥアルコム・インコーポレイテッド 高帯域バーストの抑制のためのシステム、方法、および装置
US7342525B2 (en) * 2005-12-23 2008-03-11 Cirrus Logic, Inc. Sample rate conversion combined with DSM
EP2077550B8 (fr) * 2008-01-04 2012-03-14 Dolby International AB Encodeur audio et décodeur
EP2676263B1 (fr) * 2011-02-16 2016-06-01 Dolby Laboratories Licensing Corporation Procédé de configuration de filtres

Patent Citations (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5974380A (en) * 1995-12-01 1999-10-26 Digital Theater Systems, Inc. Multi-channel audio decoder
US6975254B1 (en) * 1998-12-28 2005-12-13 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Methods and devices for coding or decoding an audio signal or bit stream
US6493664B1 (en) * 1999-04-05 2002-12-10 Hughes Electronics Corporation Spectral magnitude modeling and quantization in a frequency domain interpolative speech codec system
US20140229186A1 (en) * 2002-09-04 2014-08-14 Microsoft Corporation Entropy encoding and decoding using direct level and run-length/level context-adaptive arithmetic coding/decoding modes
US20070118362A1 (en) * 2003-12-15 2007-05-24 Hiroaki Kondo Audio compression/decompression device
US20090254783A1 (en) * 2006-05-12 2009-10-08 Jens Hirschfeld Information Signal Encoding
US20110224995A1 (en) * 2008-11-18 2011-09-15 France Telecom Coding with noise shaping in a hierarchical coder

Non-Patent Citations (7)

* Cited by examiner, † Cited by third party
Title
"7 kHz audio-coding within 64 kbit/s; G.722 (11/88)", ITU-T STANDARD, INTERNATIONAL TELECOMMUNICATION UNION, GENEVA ; CH, no. G.722 (11/88), 25 November 1988 (1988-11-25), pages 1 - 75, XP017460950 *
ATAL B S ET AL: "Adaptive predictive coding of speech signals", BELL SYSTEM TECHNICAL JOURNAL, AT AND T, SHORT HILLS, NY, US, vol. 49, no. 8, 1 October 1970 (1970-10-01), pages 1973 - 1986, XP011630204, ISSN: 0005-8580, [retrieved on 20140315], DOI: 10.1002/J.1538-7305.1970.TB04297.X *
CROCHIERE R E: "Digital signal processor: Sub-band coding", BELL SYSTEM TECHNICAL JOURNAL, AT AND T, SHORT HILLS, NY, US, vol. 60, no. 7, 1 September 1981 (1981-09-01), pages 1633 - 1653, XP011630977, ISSN: 0005-8580, [retrieved on 20140315], DOI: 10.1002/J.1538-7305.1981.TB00288.X *
DIETRICH M: "Performance and Implementation of a Robust ADPCM Algorithm for Wideband Speech coding with 64 kbit/s", PROC. INTERNATIONAL ZÜRICH SEMINAR DIGITAL COMMUNICAT,, 1 January 1984 (1984-01-01), pages 15 - 21, XP009181869 *
JAYANT N S: "Adaptive post-filtering of ADPCM speech", BELL SYSTEM TECHNICAL JOURNAL, AT AND T, SHORT HILLS, NY, US, vol. 60, no. 5, 1 May 1981 (1981-05-01), pages 707 - 717, XP011630513, ISSN: 0005-8580, [retrieved on 20140315], DOI: 10.1002/J.1538-7305.1981.TB00258.X *
MARTIN HOLTERS ET AL: "Delay-free lossy audio coding using shelving pre and post-filters", ACOUSTICS, SPEECH AND SIGNAL PROCESSING, 2008. ICASSP 2008. IEEE INTERNATIONAL CONFERENCE ON, IEEE, PISCATAWAY, NJ, USA, 31 March 2008 (2008-03-31), pages 209 - 212, XP031250525, ISBN: 978-1-4244-1483-3 *
RAMAMOORTHY V ET AL: "ENHANCEMENT OF ADPCM SPEECH CODING WITH BACKWARD-ADAPTIVE ALGORITHMS FOR POSTFILTERING AND NOISE FEEDBACK", ACM TRANSACTIONS ON COMPUTER SYSTEMS (TOCS), ASSOCIATION FOR COMPUTING MACHINERY, INC, US, vol. 6, no. 2, 1 February 1988 (1988-02-01), pages 364 - 382, XP001068182, ISSN: 0734-2071 *

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