EP1158495B1 - Vorrichtung und Verfahren zur Breitbandcodierung von Sprachsignalen - Google Patents

Vorrichtung und Verfahren zur Breitbandcodierung von Sprachsignalen Download PDF

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Publication number
EP1158495B1
EP1158495B1 EP01000172A EP01000172A EP1158495B1 EP 1158495 B1 EP1158495 B1 EP 1158495B1 EP 01000172 A EP01000172 A EP 01000172A EP 01000172 A EP01000172 A EP 01000172A EP 1158495 B1 EP1158495 B1 EP 1158495B1
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EP
European Patent Office
Prior art keywords
speech
lowband
highband
khz
wideband
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EP01000172A
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English (en)
French (fr)
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EP1158495A3 (de
EP1158495A2 (de
Inventor
Alan V Mccree
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Texas Instruments Inc
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Texas Instruments Inc
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Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
    • 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
    • G10L19/0204—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition

Definitions

  • the present invention relates to electronic devices, and, more particularly, to speech coding, transmission, storage, and decoding/synthesis methods and systems.
  • the performance of digital speech systems using low bit rates has become increasingly important with current and foreseeable digital communications.
  • Both dedicated channel and packetized-over-network (VoIP) transmission benefit from compression of speech signals.
  • the widely-used linear prediction (LP) digital speech coding compression method models the vocal tract as a time-varying filter and a time-varying excitation of the filter to mimic human speech.
  • M the order of the linear prediction filter, is taken to be about 10-12; the sampling rate to form the samples s(n) is typically taken to be 8 kHz (the same as the public switched telephone network (PSTN) sampling for digital transmission); and the number of samples ⁇ s(n) ⁇ in a frame is often 80 or 160 (10 or 20 ms frames).
  • Various windowing operations may be applied to the samples of the input speech frame.
  • ⁇ r(n) 2 yields the ⁇ a(j) ⁇ which furnish the best linear prediction.
  • the coefficients ⁇ a(j) ⁇ may be converted to line spectral frequencies (LSFs) for quantization and transmission or storage.
  • the ⁇ r(n) ⁇ form the LP residual for the frame, and ideally LP residual would be the excitation for the synthesis filter 1/A(z) where A(z) is the transfer function of equation (1).
  • the LP residual is not available at the decoder; thus the task of the encoder is to represent the LP residual so that the decoder can generate an LP excitation from the encoded parameters.
  • the LP compression approach basically only transmits/stores updates for the (quantized) filter coefficients, the (quantized) residual (waveform or parameters such as pitch), and the (quantized) gain.
  • a receiver regenerates the speech with the same perceptual characteristics as the input speech.
  • Figure 9 shows the blocks in an LP system. Periodic updating of the quantized items requires fewer bits than direct representation of the speech signal, so a reasonable LP coder can operate at bits rates as low as 2-3 kb/s (kilobits per second).
  • the ITU standard G.729 Annex E with a bit rate of 11.8 kb/s uses LP analysis with codebook excitation (CELP) to compress voiceband speech and has performance comparable to the 64 kb/s PCM used for PSTN digital transmission.
  • CELP codebook excitation
  • Another approach uses split-band CELP or MPLPC by coding a 4-8 kHz highband separately from the 0-4 kHz lowband and with fewer bits allocated to the highband; see Drogo de Jacovo et al, Some Experiments of 7 kHz Audio Coding at 16 kbit/s, IEEE ICASSP 1989, pp.192-195. Similarly, Tucker, Low Bit-Rate Frequency Extension Coding, IEE Colloquium on Audio and Music Technology 1998, pp.3/1-3/5, provides standard coding of the lowband 0-4 kHz plus codes the 4-8 kHz highband speech only for unvoiced frames (as determined in the lowband) and uses an LP filter of order 2-4 with noise excitation.
  • these approaches suffer from either too high a bit rate or too low a quality.
  • the present invention provides a method of wideband speech encoding, comprising: (a) partitioning a frame of digital speech into a lowband and a highband; (b) encoding said lowband; (c) encoding said highband using a linear prediction excitation from noise modulated by a portion of said lowband; and (d) combining said encoded lowband and said encoded highband to form an encoded wideband speech.
  • a method of wideband speech decoding comprising: (a) decoding a first portion of an input signal as a lowband speech signal; (b) decoding a second portion of an input signal as a noise-modulated excitation of a linear prediction encoding wherein said noise modulated excitation is noise modulated by a portion of the results of said decoding as a lowband speech signal of preceding step (a); and (c) combining the results of foregoing steps (a) and (b) to form a decoded wideband speech signal.
  • a wideband speech encoder comprising: (a) a lowband filter and a highband filter for digital speech; (b) a first encoder with input from said lowband filter; (c) a second encoder with input from said highband filter and said lowband filter, said second encoder using an excitation from noise modulated by a portion of output from said lowband filter; and (d) a combiner for the outputs of said first encoder and said second encoder to output encoded wideband speech.
  • a wideband speech decoder comprising: (a) a first speech decoder with an input for encoded narrowband speech; (b) a second speech decoder with an input for encoded highband speech and an input for the output of said first speech decoder, said second speech decoder using excitation of noise modulated by a portion of the output of said first speech decoder; and (c) a combiner for the outputs of said first and second speech decoders to output decoded wideband speech.
  • the preferred embodiment systems include preferred embodiment encoders and decoders that process a wideband speech frame as the sum of a lowband signal and a highband signal in which the lowband signal has standalone speech encoding/decoding and the highband signal has encoding/decoding incorporating information from the lowband signal to modulate a noise excitation. This allows for a minimal number of bits to sufficiently encode the highband and yields an embedded coder.
  • Figure 1a shows in functional block format a first preferred embodiment system for wideband speech encoding, transmission (storage), and decoding including first preferred embodiment encoders and decoders.
  • the encoders and decoders use CELP lowband encoding and decoding plus a highband encoding and decoding incorporating information from the (decoded) lowband for modulation of a noise excitation with LP coding.
  • first preferred embodiment encoders proceed as follows.
  • the baseband of the decimated highband has a reversed spectrum because the baseband is an aliased image; see Figure 3b.
  • encode the first baseband (decimated lowband) signal with a (standard) narrowband speech coder.
  • the ITU G.729 standard 8 kb/s uses 18 bits for quantized LP coefficients (three codebooks) per 10 ms (80 samples) frame, 14 bits for pitch delay (adaptive codebook), 34 bits for delayed excitation differential (fixed codebook), and 14 bits for gains.
  • Figures 4a-4b show block diagrams of the encoder and decoder.
  • G.729 Annex E provides higher quality with a higher bit rate (11.8 kb/s).
  • Decoding reverses the encoding process by separating the highband and lowband code, using information from the decoded lowband to help decode the highband, and adding the decoded highband to the decoded lowband speech to synthesize wideband speech. See Figure 1c.
  • This split-band approach allows most of the code bits to be allocated to the lowband; for example, the lowband may consume 11.8 kb/s and the highband may add 2.2 kb/s for a total of 14 kb/s.
  • Figures 2a-2b illustrate the typical magnitudes of voiced and unvoiced speech, respectively, as functions of frequency over the range 0-8 kHz.
  • the bulk of the energy in voiced speech resides in the 0-3 kHz band.
  • the pitch structure (the fundamental frequency is about 125 Hz in Figure 2a) clearly appears in the range 0-3.5 kHz and persists (although jumbled) at higher frequencies.
  • the perceptual critical bandwidth at higher frequencies is roughly 10% of a band center frequency, so the individual pitch harmonics become indistinguishable and should require fewer bits for inclusion in a highband code.
  • Figure 2b shows unvoiced-speech energy peaks in the 3.5-6.5 kHz band.
  • the precise character of this highband signal contains little perceptual information.
  • the higher band (above 4 kHz) should require fewer bits to encode than the lower band (0-4 kHz).
  • This underlies the preferred embodiment methods of partitioning wideband (0-8 kHz) speech into a lowband (0-4 kHz) and a highband (4-8 kHz), recognizing that the lowband may be encoded by any convenient narrowband coder, and separately coding the highband with a relatively small number of bits as described in the following sections.
  • Figure 1b illustrates the flow of a first preferred embodiment speech coder which encodes at 14 kb/s with the following steps.
  • a first preferred embodiment decoding method essentially reverses the encoding steps for a bitstream encoded by the first preferred embodiment method.
  • a coded frame in the bitstream For a coded frame in the bitstream:
  • FIGS 8-9 show in functional block form preferred embodiment systems that use the preferred embodiment encoding and decoding.
  • the encoding and decoding can be performed with digital signal processors (DSPs) or general purpose programmable processors or application specific circuitry or systems on a chip such as both a DSP and RISC processor on the same chip with the RISC processor controlling.
  • Codebooks would be stored in memory at both the encoder and decoder, and a stored program in an onboard ROM or external flash EEPROM for a DSP or programmable processor could perform the signal processing.
  • Analog-to-digital converters and digital-to-analog converters provide coupling to the real world, and modulators and demodulators (plus antennas for air interfaces) provide coupling for transmission waveforms.
  • the encoded speech can be packetized and transmitted over networks such as the Internet.
  • Second preferred embodiment coders and decoders follow the first preferred embodiment coders and decoders and partition the sampled input into a lowband and a highband, downsample, and apply a narrowband coder to the lowband.
  • the second preferred embodiments vary the encoding of the highband with modulated noise-excited LP by deriving the modulation from the envelope of lbdh(m) rather than its absolute value. In particular, find the envelope en(m) of lbdh(m) by lowpass (0-1 kHz) filtering the absolute value
  • Figure 7 illustrates en(m) for the voiced speech of Figure 6 in the time domain.
  • the preferred embodiments may be modified in various ways while retaining the features of separately coding a lowband from a wideband signal and using information from the lowband to help encode the highband (remainder of the wideband) and/or using spectrum reversal for decimated highband LP coefficient quantization in order to obtain efficiency comparable to that for the lowband LP coefficient quantization.
  • the upper (2.8-3.8 kHz) portion of the lowband (0-4 kHz) could be replaced by some other portion(s) of the lowband for use as a modulation for the highband excitation.
  • the highband encoder/decoder may have its own LP analysis and quantization, so the spectral reversal would not be required; the wideband may be partitioned into a lowband plus two or more highbands; the lowband coder could be a parametric or even non-LP coder and a highband coder could be a waveform coder; and so forth.
  • the scope of the invention is hereby only limited by the appended claims.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Quality & Reliability (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Analogue/Digital Conversion (AREA)

Claims (4)

  1. Verfahren für Breitbandsprachkodierung, das folgendes umfasst:
    (a) Aufteilen eines Rahmens digitaler Sprache in ein Niedrigband und ein Hochband;
    (b) Kodierung des Niedrigbands
    (c) Kodierung des Hochbands unter Anwendung einer linearen Vorhersageerregung aus einem Rauschen, das von einem Anteil des Niedrigbands moduliert wird; und
    (d) Kombination des kodierten Niedrigbands mit dem kodierten Hochband, um eine kodierte Breitbandsprache zu schaffen.
  2. Verfahren für Breitbandsprachdekodierung, das folgendes umfasst:
    (a) Dekodierung eines ersten Anteils eines Eingangssignals als Hochband-Sprachsignal;
    (b) Dekodierung eines zweiten Anteils eines Eingangssignals als eine geräuschmodulierte Erregung einer linearen Vorhersagekodierung, wobei die Geräuschmodulation der geräuschmodulierten Erregung durch einen Anteil der Ergebnisse des Dekodierens als ein Niedrigband-Sprachsignal aus dem vorhergehenden Schritt (a) stattfindet; und
    (c) Kombination der Ergebnisse der vorhergehenden Schritte (a) und (b), um ein dekodiertes Breitbandsprachsignal zu schaffen.
  3. Breitbandsprachkodiereinrichtung, die folgendes umfasst:
    (a) ein Niedrigbandfilter und ein Hochbandfilter für digitale Sprache;
    (b) eine erste Kodiereinrichtung, die Eingabesignale vom Niedrigbandfilter erhält;
    (c) eine zweite Kodiereinrichtung, die Eingabesignale vom Hochbandfilter und vom Niedrigbandfilter erhält, wobei die zweite Kodiereinrichtung eine Erregung aus einem Rauschen verwendet, das von einem Anteil der Ausgabe des Niedrigbandfilters moduliert wird; und
    (d) einen Kombinator, damit die Ausgänge der ersten Kodiereinrichtung und der zweiten Kodiereinrichtung kodierte Breitbandsprache ausgeben.
  4. Breitbandsprachdekodiereinrichtung, die folgendes umfasst:
    (a) einen ersten Sprachdekodierer mit einem Eingang für kodierte Schmalbandsprache;
    (b) einen zweiten Sprachdekodierer, mit einem Eingang für kodierte Hochbandsprache und einem Eingang für die Ausgabe des ersten Sprachdekodierers, wobei der zweite Sprachdekodierer eine Erregung aus einem Rauschen verwendet, das von einem Anteil der Ausgabe des ersten Sprachdekodierers moduliert wird; und
    (c) einen Kombinator, damit die Ausgänge des ersten Sprachdekodierers und des zweiten Sprachdekodierers dekodierte Breitbandsprache ausgeben.
EP01000172A 2000-05-22 2001-05-22 Vorrichtung und Verfahren zur Breitbandcodierung von Sprachsignalen Expired - Lifetime EP1158495B1 (de)

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EP04100553A Division-Into EP1431962B1 (de) 2000-05-22 2001-05-22 Vorrichtung und Verfahren zur Breitbandcodierung von Sprachsignalen

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* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2004085604A (ja) * 2002-08-22 2004-03-18 Alpine Electronics Inc デジタルオーディオ信号伝送方法及びオーディオシステム
WO2004090870A1 (ja) 2003-04-04 2004-10-21 Kabushiki Kaisha Toshiba 広帯域音声を符号化または復号化するための方法及び装置
US7848921B2 (en) 2004-08-31 2010-12-07 Panasonic Corporation Low-frequency-band component and high-frequency-band audio encoding/decoding apparatus, and communication apparatus thereof
BRPI0515453A (pt) 2004-09-17 2008-07-22 Matsushita Electric Ind Co Ltd aparelho de codificação escalável, aparelho de decodificação escalável, método de codificação escalável método de decodificação escalável, aparelho de terminal de comunicação, e aparelho de estação de base
US7783480B2 (en) 2004-09-17 2010-08-24 Panasonic Corporation Audio encoding apparatus, audio decoding apparatus, communication apparatus and audio encoding method
KR100721537B1 (ko) 2004-12-08 2007-05-23 한국전자통신연구원 광대역 음성 부호화기의 고대역 음성 부호화 장치 및 그방법
KR100707174B1 (ko) 2004-12-31 2007-04-13 삼성전자주식회사 광대역 음성 부호화 및 복호화 시스템에서 고대역 음성부호화 및 복호화 장치와 그 방법
CA2603229C (en) 2005-04-01 2012-07-31 Qualcomm Incorporated Method and apparatus for split-band encoding of speech signals
DK1875463T3 (en) 2005-04-22 2019-01-28 Qualcomm Inc SYSTEMS, PROCEDURES AND APPARATUS FOR AMPLIFIER FACTOR GLOSSARY
JP5030789B2 (ja) * 2005-11-30 2012-09-19 パナソニック株式会社 サブバンド符号化装置およびサブバンド符号化方法
US8301281B2 (en) 2006-12-25 2012-10-30 Kyushu Institute Of Technology High-frequency signal interpolation apparatus and high-frequency signal interpolation method
WO2012055016A1 (en) * 2010-10-25 2012-05-03 Voiceage Corporation Coding generic audio signals at low bitrates and low delay

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EP1158495A3 (de) 2002-01-30
DE60102975T2 (de) 2005-05-12
ATE265732T1 (de) 2004-05-15
DE60102975D1 (de) 2004-06-03
EP1158495A2 (de) 2001-11-28
JP2001337700A (ja) 2001-12-07
DE60118627T2 (de) 2007-01-11
DE60118627D1 (de) 2006-05-18

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