EP0655731A2 - Rauschunterdrückungseinrichtung zur Vorverarbeitung und/oder Nachbearbeitung von Sprachsignalen - Google Patents

Rauschunterdrückungseinrichtung zur Vorverarbeitung und/oder Nachbearbeitung von Sprachsignalen Download PDF

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Publication number
EP0655731A2
EP0655731A2 EP94118782A EP94118782A EP0655731A2 EP 0655731 A2 EP0655731 A2 EP 0655731A2 EP 94118782 A EP94118782 A EP 94118782A EP 94118782 A EP94118782 A EP 94118782A EP 0655731 A2 EP0655731 A2 EP 0655731A2
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Prior art keywords
signal
speech
noise
feature parameter
produce
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EP94118782A
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English (en)
French (fr)
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EP0655731B1 (de
EP0655731A3 (de
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Kazunori C/O Nec Corporation Ozawa
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NEC Corp
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NEC Corp
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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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise 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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
    • G10L21/0364—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude for improving intelligibility

Definitions

  • This invention relates to a noise suppressor for use in suppressing a noise signal from a speech signal.
  • a speech signal is subjected to pre-processing before the speech signal is encoded into a sequence of encoded signals.
  • pre-processing has been made to judge either a speech duration or a non-speech duration, in an article which is contributed by J.F. Lynch, Jr. et al to IEEE and which is entitled "SPEECH/SILENCE SEGMENTATION FOR REAL-TIME CODING VIA RULE BASED ADAPTIVE ENDPOINT DETECTION" (Proceedings ICASSP, pages 1348-1351, 1987).
  • description is made only about detection between the speech duration and the non-speech duration but is not made about suppressing a noise signal from the speech signal during the pre-processing.
  • speech encoding is usually carried out in connection not only with the spectrum but also with a phase component of the speech signal. This shows that a noise component can not be removed which is included in the phase component in the above-mentioned method.
  • the spectrum subtraction is disadvantageous in that the noise component can not be completely suppressed from the speech signal.
  • the spectrum subtraction can not be applied on post-processing which is carried out after the encoded signal sequence is decoded into a sequence of decoded signals.
  • a noise suppressor is operable to carry out pre-processing in relation to a speech signal.
  • the noise suppressor is supplied with an internal input signal which includes both the speech signal and a noise signal to produce an output signal substantially free from the noise signal.
  • the speech signal is specified by a sound source.
  • the noise suppressor comprises feature parameter calculating means supplied with said internal input signal for calculating a feature parameter specifying a feature of the speech signal to produce a feature parameter signal representative of said feature parameter and noise suppressing means coupled to the feature parameter calculating means for suppressing the noise signal from the internal input signal to produce the output signal.
  • the noise suppressing means comprises filter means supplied with the feature parameter signal and the internal input signal for filtering the internal input signal to produce a filtered signal which is dependent on the feature parameter and which specifies the sound source, a suppression unit coupled to the filter means for suppressing the noise signal from the filtered signal by estimating the noise signal to produce a noise-suppressed signal, output means for producing the noise-suppressed signal as the output signal.
  • a noise suppressor is operable to carry out post-processing of a speech signal and responds to a feature parameter signal specifying the speech signal and a sound source signal representative of a sound source of the speech signal to suppress a noise signal from the speech signal and to produce an output signal substantially free from the noise signal.
  • the speech signal is divisible into a speech duration and a non-speech duration.
  • the noise suppressor comprises a noise suppressing circuit for suppressing the noise signal from the sound source signal with reference to the feature parameter signal to produce a noise-suppressed signal and means for producing the noise-suppressed signal as the output signal.
  • a speech signal is given in the form of a sequence of digital speech signals to be subjected to pre-processing and post-processing to suppress a noise signal from the speech signal.
  • the pre-processing is carried out in response to an input signal specified by the digital speech signal sequence which is not encoded yet while the post-processing is carried out in response to an input signal specified by the digital speech signal sequence which is already decoded. Therefore, it is noted that the terms "digital speech signal sequence” and "input signal” may be used in two different meanings hereinunder so as to include both the pre-processing and the post-processing.
  • the input signal includes the speech signal (namely, the digital speech signal sequence) and the noise signal and may be therefore considered as a combination of the digital speech signal sequence and the noise signal.
  • feature parameters are extracted from the input signal and may be, for example, selected one or ones of spectrum parameters representative of features of a spectrum in the input signal, pitch prediction gains representative of periodicity of the input signal, and the like.
  • the feature parameters are used to determine either a speech duration or a non-speech duration by comparing the feature parameters with a threshold level.
  • a preliminary sound source signal which specifies a sound source is obtained by the use of the input signal and the feature parameters on the pre-processing and the post-processing. Specifically, the preliminary sound source signal appears in the form of an error signal which is produced on the pre-processing by allowing the input signal to pass through an inverse filter controlled by the feature parameters.
  • the preliminary sound source signal appears in the form of a decoder output signal or a sequence of decoded signals which is decoded by the use of the feature parameters.
  • the speech signal has an amplitude greater than the noise signal in the preliminary sound source signal, it is possible to suppress the noise signal alone by comparing an amplitude of the preliminary sound source signal with a predetermined threshold level and to therefore attain a noise-suppressed signal.
  • the noise-suppressed signal is reproduced by the use of the feature parameters into a noise-free output signal on the pre-processing or is produced as a noise-free decoded signal on the post-processing.
  • the noise-free output signal may be encoded by an encoder after the pre-processing while the noise-free decoded signal may be converted into an audio signal after the post-processing.
  • Noise suppression may be carried out only within a selected one of the speech duration or the non-speech duration or within both the speech duration and the non-speech duration.
  • this invention enables to suppress the noise signal on a waveform by the use of the feature parameters and is applicable to both the pre-processing and the post-processing.
  • a noise suppressor is applicable to the pre-processing and is therefore supplied through an input terminal 10 with an input signal IN which includes a speech signal and a noise signal superposed on the speech signal.
  • the speech signal is given in the form of a sequence of digital speech signals.
  • the input signal IN is given to a frame division circuit 11 and is divided by the frame division circuit 11 into a plurality of frames each of which has a length of, for example, 40 milliseconds.
  • Each frame is further subdivided by a subframe division circuit 12 into a plurality of subframes each of which has a length of, for example, eight milliseconds.
  • the input signal IN is divided into the subframes, as mentioned above, and is sent in the form of a divided input signal sequence x(n) either at every frame or at every subframe to a feature parameter calculator 15 on one hand and to a noise suppression circuit 20 on the other hand.
  • the divided input signal sequence x(n) may be referred to as an internal input signal.
  • the feature parameter calculator 15 is supplied with the internal input signal x(n) at every subframe.
  • the feature parameter calculator 15 at first places a window to extract a piece of the internal input signal x(n) in relation to each subframe.
  • the window is longer than each subframe length and may be, for example, 24 milliseconds.
  • the feature parameter calculator 15 calculates, as feature parameters, spectrum parameters indicative of features of a spectrum in the input signal, pitch prediction gains indicative of periodicity of the speech signal, and an average amplitude in each subframe. In this event, average power may be calculated in the feature parameter calculator 15. Such calculations of the feature parameters are known in the art and will not be described any longer. In any event, the feature parameters are produced as feature parameter signals from the feature parameter calculator 15.
  • the feature parameter calculator 15 shown in Fig. 1 calculates the spectrum parameters of a predetermined order which may be, for example, a tenth order.
  • a i linear prediction coefficients
  • the Burg analysis is used to calculate the linear prediction coefficients.
  • the Burg analysis is described in detail in a book (pages 82 to 87) which is written by Nakamizo et al and which is titled "Signal Analysis and System Identification” published by Corona Company Ltd, Tokyo, in 1988. Accordingly, description will be omitted from the instant specification as regards the Burg analysis.
  • the linear prediction coefficients may be also calculated by the use of a covariance method or a correlation method.
  • the pitch prediction gains are also calculated in the feature parameter calculator 15.
  • the pitch prediction gains are represented by P g and are given by: where T is a delay time representative of a pitch period; n, a sample number; and N, a maximum sample number.
  • the pitch prediction gains P g can be simply calculated by the use of the following equation:
  • the average amplitude is represented by R and is given by:
  • circuits for calculating the above-mentioned linear prediction coefficients, the pitch prediction gains P g , and the average amplitude R by a combination of conventional circuit elements. Accordingly, specific circuits for calculating the linear prediction coefficients, the pitch prediction gains P g , and the average amplitude will not be described later.
  • the feature parameter calculator 15 supplies a speech detection circuit 25 and the noise suppression circuit 20 with the feature parameter signals representative of the feature parameters, as mentioned above.
  • the speech detection circuit 25 detects or determines either the speech duration or the non-speech duration of the speech signal in response to at least one of the feature parameters. To this end, a wide variety of methods can be applied to determine the speech duration or the non-speech duration.
  • the illustrated speech detection circuit 25 at first smooths the pitch prediction gains P g and the average amplitude R to obtain smoothed pitch prediction gains P g ' and a smoothed average amplitude R' and thereafter compares the smoothed pitch prediction gains P g ' and the smoothed average amplitude R' with first and second threshold values Th1 and Th2, respectively.
  • the speech detection circuit 25 judges that the non-speech duration lasts in the internal input signal x(n). Otherwise, the speech detection circuit 25 judges that the speech duration lasts in the internal input signal x(n). Thus, the non-speech and the speech durations are detected by the speech detection circuit 25.
  • the first and the second threshold values Th1 and Th2 may be invariable or variable with time.
  • the speech detection circuit 25 comprises a calculation circuit for calculating the smoothed values (namely, the smoothed pitch prediction gains P g ' and the smoothed average amplitude R') in accordance with Equation 4 and a comparator unit for comparing the smoothed values with the first and the second threshold values Th1 and Th2.
  • the illustrated speech detection circuit 25 can produce the smoothed average amplitude R' at every frame or at every subframe and a detection signal DT representative of either the speech or the non-speech duration at every frame or at every subframe.
  • the smoothed average amplitude R' is delivered to a memory circuit 30 while the detection signal DT is sent to the noise suppression circuit 20.
  • the noise suppression circuit 20 is operable to suppress the noise signal within at least one of the speech and the non-speech durations.
  • the noise suppression circuit 20 comprises an inverse filter 201 supplied with the internal input signal x(n) from the input terminal 10 through the frame and the subframe division circuits 11 and 12.
  • the feature parameters a i are also supplied from the feature parameter calculator 15 to the inverse filter 201.
  • the inverse filter 201 carries out an inverse filtering operation to produce an inverse-filtered signal e(n) which may be called a preliminary sound source signal because the inverse-filtered signal e(n) specifies a sound source.
  • the inverse-filtered signal e(n) is given by: where P represents an order of the inverse filter 201.
  • the inverse-filtered signal e(n) is dependent on the feature parameters and specifies the sound source.
  • the inverse-filtered signal e(n) includes a speech signal component and a noise signal component superposed on the speech signal component and appears in the form of a continuous signal.
  • the inverse filter 201 may be simply called a filter circuit.
  • the inverse-filtered signal e(n) is specified by a comparatively large amplitude pulse within a portion of the speech signal component appearing in the speech duration because the speech signal has a pitch.
  • the inverse-filtered signal e(n) exhibits a comparatively small amplitude within a portion of the noise signal.
  • the noise suppression circuit 20 illustrated in Fig. 2 comprises a threshold value calculation circuit 202 supplied with the smoothed average amplitude R' which is calculated by the feature parameter calculator 15 in accordance with Equation 4 and which is memorized into the memory circuit 30.
  • the threhold value TH1 is determined by the average amplitude R memorized in the memory circuit 30.
  • the inverse-filtered signal e(n) and the threshold value signal are sent to a suppressor unit 203 which is also given the detection signal DT from the speech detection circuit 25.
  • the suppressor unit 203 is put into an active state or into an inactive state in response to the detection signal DT. In this event, the suppressor unit 203 may suppress the noise signal within at least one of the speech duration and the non-speech duration. In the illustrated example, it is assumed that the suppressor unit 203 is put into the active state within the non-speech duration in response to the detection signal DT, although the suppressor unit 203 may be put into the active state within the speech duration.
  • the suppressor unit 203 compares the inverse-filtered signal e(n) with the threshold value signal.
  • the suppressor unit 203 attenuates the inverse-filtered signal e(n) by a predetermined amount or renders the inverse-filtered signal e(n) into zero when the inverse-filtered signal e(n) is smaller than the threshold value TH1.
  • the suppressor unit 203 produces a noise-suppressed signal e' specified by: where K is greater than zero and smaller than unity.
  • a combination of the threshold value calculation circuit 202 and the suppressor unit 203 serves to suppress the noise signal included in the inverse-filtered signal e(n) and to produce the noise-suppressed signal e'(n) and may be collectively called a noise suppression portion.
  • the noise-suppressed signal e'(n) is sent to a reproduction circuit 204 together with the feature parameters a i .
  • the reproduction circuit 204 reproduces the noise-suppressed signal e'(n) into a noise-suppressed speech signal x'(n) with reference to the feature parameters ai.
  • the noise-suppressed speech signal x' is given by:
  • the noise-supressed speech signal x'(n) is delivered through an output terminal 35 of the noise suppression circuit 20 to an encoder (not shown) to be encoded.
  • the noise-suppressed speech signal x'(n) is produced during the pre-processing prior to the encoding. Since the noise-suppression is carried out with reference to the feature parameters of the input signal IN, a phase component of the noise signal can also be suppressed in the above-mentioned example.
  • a noise suppressor (depicted at 40) according to a second embodiment of this invention is operable to carry out post-processing after decoding.
  • the illustrated noise processor 40 is connected to a decoder 45 which is supplied as a decoder input signal or an input signal DIN with feature parameters of a speech signal and an index signal related to a sound source.
  • the decoder 45 itself may be similar to that known in the art and produces a sequence of decoded sound source signals v(n) representative of a sound source together with the feature parameters and the index signal, in a known manner.
  • the decoded sound source signal sequence v(n) and the feature parameters and the index signal are sent to the noise suppressor 40.
  • the decoded sound source signal sequence v(n) is given to a noise suppression circuit which is depicted at 50 and which is operable in a manner to be described later in detail.
  • the illustrated noise suppressor 40 comprises a speech detection circuit 25' and a memory circuit 30' which may be similar to those illustrated in Fig. 1, respectively. From this fact, it is readily understood that the speech detection circuit 30' is operated in response to the feature parameters, such as the spectrum parameters, the pitch prediction gains P g , and the average amplitude R, to detect either the speech duration or the non-speech duration.
  • the speech detection circuit 30' supplies the noise suppression circuit 50 with a detection signal DT' indicative of either the speech duration or the non-speech duration.
  • the speech detection circuit 25' calculates the smoothed average amplitude R' which is stored in the memory circuit 30'.
  • the noise suppression circuit 50 comprises a threshold calculator 501 supplied with the smoothed average amplitude R' to calculate a threshold value signal representative of a threshold value TH2, like in the threshold value calculation circuit 202.
  • the threshold value signal is given to the suppressor unit 502 together with the detection signal DT'.
  • the suppressor unit 502 is put into an active state within at least one of the speech and the non-speech durations.
  • the illustrated suppressor unit 502 becomes active only within the non-speech duration, like in the suppressor unit 203.
  • the suppressor unit 502 produces a sequence of noise-suppressed sound source signals v'(n) given by: where K is identical with K shown in Equation 7.
  • the threshold value TH2 may be equal to that of Equation 7.
  • the noise-suppressed sound source signals v'(n) are sent to a speech reproducing circuit 52 which is supplied with the feature parameters from the decoder 45.
  • the speech reproducing circuit 52 reproduces the noise-suppressed sound signals into a reproduced speech signal with reference to the feature parameters in a known manner.
  • the reproduced speech signal is delivered to a loudspeaker or the like.
  • the noise suppressor according to this invention can be used in post-processing the decoded sound source signals DIN in the above-mentioned manner.
  • the feature parameters may not be always restricted to the linear prediction coefficients but may be any other parameters known in the art.
  • the speech detection circuit 25 or 25' may be operated in a manner different from that illustrated in Figs. 1 and 3.
  • the post-processing can be carried out to suppress the noise signal even when the feature parameters are not transmitted from a transmitter and are not received by the decoder 45 (Fig. 3).
  • the speech signal is once reproduced by a receiver to form a reproduced speech waveform and to thereafter calculate feature parameters from the reproduced speech waveform in the manner mentioned in conjunction with Fig. 1.
  • the calculated feature parameters can be used to suppress the noise signal in the above-mentioned manner.
  • the noise suppression is possible during both the pre-processing and the post-processing of the speech signal. Moreover, it is also possible to suppress not only the noise signal appearing within the non-speech duration but also a non-speech signal superposed on the speech signal appearing within the speech duration. Such suppression can be accomplished on the waveform.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (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)
  • Noise Elimination (AREA)
  • Filters That Use Time-Delay Elements (AREA)
EP19940118782 1993-11-29 1994-11-29 Rauschunterdrückungseinrichtung zur Vorverarbeitung und/oder Nachbearbeitung von Sprachsignalen Expired - Lifetime EP0655731B1 (de)

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JP29717693 1993-11-29
JP29717693A JP2739811B2 (ja) 1993-11-29 1993-11-29 雑音抑圧方式

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EP0655731A2 true EP0655731A2 (de) 1995-05-31
EP0655731A3 EP0655731A3 (de) 1997-05-28
EP0655731B1 EP0655731B1 (de) 2000-03-29

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Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0751491A2 (de) * 1995-06-30 1997-01-02 Sony Corporation Verfahren zur Rauschverminderung in einem Sprachsignal
WO2001082294A1 (en) * 2000-04-24 2001-11-01 Telefonaktiebolaget Lm Ericsson (Publ) System and method for distributed noise suppression
EP2229675A1 (de) * 2007-12-06 2010-09-22 Electronics and Telecommunications Research Institute Vorrichtung und verfahren zur erhöhung der qualität eines sprachcodecs
EP2603914A2 (de) * 2010-08-11 2013-06-19 Bone Tone Communications Ltd. Hintergrundklangunterdrückung zur verwendung für privatsphärensicherung und personalisierung
US10755731B2 (en) 2016-09-08 2020-08-25 Fujitsu Limited Apparatus, method, and non-transitory computer-readable storage medium for storing program for utterance section detection

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0438174A2 (de) * 1990-01-18 1991-07-24 Matsushita Electric Industrial Co., Ltd. Einrichtung zur Signalverarbeitung
EP0459364A1 (de) * 1990-05-28 1991-12-04 Matsushita Electric Industrial Co., Ltd. Geräuschsignalvorhersagevorrichtung
JPH05188994A (ja) * 1992-01-07 1993-07-30 Sony Corp 騒音抑圧装置

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0438174A2 (de) * 1990-01-18 1991-07-24 Matsushita Electric Industrial Co., Ltd. Einrichtung zur Signalverarbeitung
EP0459364A1 (de) * 1990-05-28 1991-12-04 Matsushita Electric Industrial Co., Ltd. Geräuschsignalvorhersagevorrichtung
JPH05188994A (ja) * 1992-01-07 1993-07-30 Sony Corp 騒音抑圧装置

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
PALIWAL K K: "Speech enhancement using multipulse excited linear prediction system" , ICASSP 86 PROCEEDINGS. IEEE-IECEJ-ASJ INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (CAT. NO.86CH2243-4), TOKYO, JAPAN, 7-11 APRIL 1986 , 1986, NEW YORK, NY, USA, IEEE, USA, PAGE(S) 101 - 104 VOL.1 XP002027701 * column 2, line 1 - line 10 * * figure 1 * *

Cited By (14)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0751491A2 (de) * 1995-06-30 1997-01-02 Sony Corporation Verfahren zur Rauschverminderung in einem Sprachsignal
EP0751491A3 (de) * 1995-06-30 1998-04-08 Sony Corporation Verfahren zur Rauschverminderung in einem Sprachsignal
WO2001082294A1 (en) * 2000-04-24 2001-11-01 Telefonaktiebolaget Lm Ericsson (Publ) System and method for distributed noise suppression
US7225001B1 (en) 2000-04-24 2007-05-29 Telefonaktiebolaget Lm Ericsson (Publ) System and method for distributed noise suppression
EP2560163A1 (de) * 2007-12-06 2013-02-20 Electronics and Telecommunications Research Institute Vorrichtung und Verfahren zur Erhöhung der Qualität eines Sprachcodecs
EP2229675A4 (de) * 2007-12-06 2012-03-07 Korea Electronics Telecomm Vorrichtung und verfahren zur erhöhung der qualität eines sprachcodecs
EP2229675A1 (de) * 2007-12-06 2010-09-22 Electronics and Telecommunications Research Institute Vorrichtung und verfahren zur erhöhung der qualität eines sprachcodecs
EP2560162A1 (de) * 2007-12-06 2013-02-20 Electronics and Telecommunications Research Institute Vorrichtung und Verfahren zur Erhöhung der Qualität eines Sprachcodecs
US9135925B2 (en) 2007-12-06 2015-09-15 Electronics And Telecommunications Research Institute Apparatus and method of enhancing quality of speech codec
US9135926B2 (en) 2007-12-06 2015-09-15 Electronics And Telecommunications Research Institute Apparatus and method of enhancing quality of speech codec
US9142222B2 (en) 2007-12-06 2015-09-22 Electronics And Telecommunications Research Institute Apparatus and method of enhancing quality of speech codec
EP2603914A2 (de) * 2010-08-11 2013-06-19 Bone Tone Communications Ltd. Hintergrundklangunterdrückung zur verwendung für privatsphärensicherung und personalisierung
EP2603914A4 (de) * 2010-08-11 2014-11-19 Bone Tone Comm Ltd Hintergrundklangunterdrückung zur verwendung für privatsphärensicherung und personalisierung
US10755731B2 (en) 2016-09-08 2020-08-25 Fujitsu Limited Apparatus, method, and non-transitory computer-readable storage medium for storing program for utterance section detection

Also Published As

Publication number Publication date
DE69423703T2 (de) 2000-07-27
EP0655731B1 (de) 2000-03-29
EP0655731A3 (de) 1997-05-28
JP2739811B2 (ja) 1998-04-15
DE69423703D1 (de) 2000-05-04
JPH07152395A (ja) 1995-06-16

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