WO2009103608A1 - Verfahren und mittel zur enkodierung von hintergrundrauschinformationen - Google Patents
Verfahren und mittel zur enkodierung von hintergrundrauschinformationen Download PDFInfo
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- WO2009103608A1 WO2009103608A1 PCT/EP2009/051118 EP2009051118W WO2009103608A1 WO 2009103608 A1 WO2009103608 A1 WO 2009103608A1 EP 2009051118 W EP2009051118 W EP 2009051118W WO 2009103608 A1 WO2009103608 A1 WO 2009103608A1
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- background noise
- sid
- speech
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- 238000000034 method Methods 0.000 title claims abstract description 65
- 230000015572 biosynthetic process Effects 0.000 claims abstract description 5
- 230000005540 biological transmission Effects 0.000 description 23
- 230000008901 benefit Effects 0.000 description 5
- 206010019133 Hangover Diseases 0.000 description 4
- 230000008569 process Effects 0.000 description 4
- 230000003595 spectral effect Effects 0.000 description 4
- 238000011161 development Methods 0.000 description 3
- 230000018109 developmental process Effects 0.000 description 3
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- 238000005070 sampling Methods 0.000 description 3
- 230000002194 synthesizing effect Effects 0.000 description 3
- 230000002123 temporal effect Effects 0.000 description 3
- 238000013459 approach Methods 0.000 description 2
- 230000008859 change Effects 0.000 description 2
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- 238000007906 compression Methods 0.000 description 2
- 238000001514 detection method Methods 0.000 description 2
- 230000005284 excitation Effects 0.000 description 2
- 230000006872 improvement Effects 0.000 description 2
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Classifications
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/012—Comfort noise or silence coding
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/02—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
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/16—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
Definitions
- the invention relates to methods and means for encoding background noise information in speech signal coding methods.
- Such a limited frequency range is also provided in many speech signal coding methods for today's digital telecommunications.
- a bandwidth limitation of the analog signal Prior to a coding process, a bandwidth limitation of the analog signal is performed for this purpose.
- a codec is used which, due to the described bandwidth limitation in the frequency range between 300 Hz and 3400 Hz, is also referred to below as narrow-band speech codec (Narrow Band Speech Codec).
- the term codec is understood to mean both the coding rule for the digital coding of audio signals and the decoding rule for the decoding of data with the aim of reconstructing the audio signal.
- a narrowband speech codec is known from ITU-T Recommendation G.729.
- a transmission of a narrow-band voice signal with a data rate of 8 kbit / s is provided.
- so-called broadband speech codecs Wide Band Speech Codec
- Such an extended frequency range is, for example, between a frequency of 50 Hz and 7000 Hz.
- a wideband voice codec for example, from the ITU-T Recommendation G.729. EV known.
- coding methods for broadband speech codecs are made scalable.
- Scalability means that the transmitted coded data contain various demarcated blocks which contain the narrowband component, the broadband component and / or the full bandwidth of the coded voice signal.
- such a scalable design allows for backwards compatibility on the receiver side and, on the other hand, offers a simple possibility of adapting the data rate and the size of transmitted data frames in the transmission channel in the case of limited data transmission capacities.
- a compression of the data to be transmitted is usually provided. Compression is achieved, for example, by coding methods, for coding the
- Speech data parameters for an excitation signal and filter parameters are determined.
- the filter parameters and the excitation signal specifying parameters are then transmitted to the receiver.
- a synthetic speech signal is synthesized using the codec, which is the original one
- Speech signal is as similar as possible in terms of a subjective Horeindrucks.
- analysis-by-synthesis the determined and digitized samples are not themselves transmitted. but determined parameters that enable a receiver-side synthesis of the speech signal.
- a further measure for reducing the data transmission rate is provided by a method for discontinuous transmission (Discontinuous Transmission), which is also known in the art as DTX.
- DTX discontinuous Transmission
- the basic goal of DTX is to reduce the data transfer rate in the event of a speech break.
- a speech pause detection (Voice Activity Detection, VAD) is used, which detects falls below a certain signal level on a speech break.
- VAD Voice Activity Detection
- the receiver does not expect a complete silence during a speech break.
- a complete silence on the receiver side would lead to irritation or even the presumption of a breakdown of the connection.
- methods for generating a so-called comfort noise are applied.
- Comfort noise is noise that is synthesized to fill silence phases on the receiver's side.
- the comfort noise serves as a subjective impression of a continuing connection, without claiming the data transmission rate intended for the transmission of speech signals. In other words, less effort is required to code the speech data for the transmitter-side coding of the noise. For a receiver-side still perceived as realistic synthesizing the comfort noise data are transmitted at a much lower data rate.
- the data transmitted here are also referred to in the art as SID (Silence Insertion Description). Codecs currently under development focus on scalable encoding of speech information.
- the result of the encoding process contains various blocks which contain the narrow-band component of the original speech signal, the broadband component or also the full bandwidth of the speech signal, eg a frequency range between 50 and 7000 Hz.
- the encoding of the background noise information occurs either over the entire bandwidth of the input noise signal or over a portion of the bandwidth of the input noise signal.
- the encoded noise signal is transmitted in the form of SID frames via the DTX method and reconstructed on the receiver side.
- synthesized comfort noise may have a different quality than the speech information synthesized on the receiver side. This has a detrimental effect on the reception of the receptionist.
- the object of the invention is to provide an improved implementation of the DTX method in scalable speech codecs.
- a basic idea of the invention is to provide the scalability known for the transmission of speech information analogously to the formation of a SID frame.
- the inventive method for encoding a SID frame for transmission of background noise information using a scalable speech signal coding method provides an encoding of a narrowband first and a broadband second portion of the background noise information.
- the encoding is usually done at the same time and in different ways. However, the encoding of a share can of course also take place with a time offset before or after an encoding of another share. Likewise, the encoding of the two components can optionally also be carried out in the same way.
- a SID frame is formed with separate regions for the first and the second component. In other words, in the SID frame, this means that a first data area receives the data for the encoded first portion, while a separate second data area receives the data for the encoded second portion.
- An essential advantage of the invention is that it can be determined on the receiver side whether a comfort noise should be based on the broadband component of the transmitted SID frames or on the basis of the narrowband component.
- This is of particular advantage for the receiver-side acoustic reception in a situation in which the transmission rate for speech information frames has been reduced so that only narrowband speech information is transmitted. Namely, as synthesized in the current state of the art, narrowband speech information in conjunction with broadband noise, this is very irritating for the receiver.
- the said reduction of the transmission rate for speech information frames can be caused for example by a high congestion of the network between transmitter and receiver.
- the much smaller SID frames are not affected by such a network bottleneck. For them, there is no compulsion to reduce their data transfer rate or their content.
- a third portion is provided in the definition of the SID frame.
- This contains encoded background noise parameters, which are encoded with an increased data rate, although the third component still contains narrow-band data (extended narrowband data or "enhanced low band").
- the frame with this third portion is a way of reproducing a noise signal in a quality enhanced in comparison with the conventional narrow-band coding method while still conforming to the G.729 standard. B to stay.
- the single FIGURE shows a structure of a SID frame according to the invention.
- Discontinuous transmission (DTX) methods implemented in presently scalable coding methods for broadband speech codecs currently do not support the scalable nature of the transmission of background noise information intended for the transmission of the speech information.
- DTX discontinuous transmission
- an encoding takes place either over the entire bandwidth of the input noise signal or over a section of the bandwidth of the input noise signal. For this reason, there is a need for improved methods.
- narrow-band speech codecs such as e.g. 3GPP AMR, ITU-T G.729 and on the other hand broadband speech codecs, such as e.g. 3GPP AMR-WB, ITU-T G.722.
- a narrowband speech codec encodes speech signals at a sampling frequency of 8 kHz with a bandwidth which is usually in the frequency range between 300 and 3400 Hz.
- a wideband speech codec encodes a speech signal having a sampling frequency of 16 kHz at a bandwidth in a frequency range between 50 and 7000 Hz.
- Some of these codecs use DTX methods, ie discontinuous transmission methods, to reduce the overall transmission rate in the communication channel.
- DTX ie discontinuous transmission methods
- SID frames are sent with the bandwidth of the SID frames being rescored with the bandwidth of the voice signal.
- a SID frame the background noise during a speech break is described.
- codecs focus on scalable coding.
- the result of the encoding process contains various blocks which contain the narrow-band component of the original speech signal, the broadband component or even the full bandwidth of the speech signal, eg a frequency range between 50 and 7000 Hz.
- the broadband component usually starts at a frequency of 4 kHz.
- the current DTX methods do not currently support the scalable nature of codecs. Instead, coding takes place either over the entire bandwidth of the input speech signal or over a section of the bandwidth of the input signal. For this reason, there is a need for improved methods.
- G.729.1 is a scalable speech codec in which the DTX method is currently not scalable across the entire bandwidth.
- the coding method can be characterized as follows during an active speech period, in contrast to a speech pause recognized as »Silent Period «:
- the speech signal is split into two parts, namely a narrowband (lowband) part and a broadband (highband) part. Both signals are sampled at a sampling frequency of 8 kHz.
- the division into a narrowband and a broadband component takes place in a special bandpass filter, which is also referred to as QMF (Quadrature Mirror Filter).
- QMF Quadrature Mirror Filter
- the narrowband portion of the speech signal is encoded at a data rate of 8 and 12 kbit / s.
- a CELP Code Excited Linear Prediction
- the narrowband component is further modified taking into account the »Transform Codec « section of G.729.1.
- the broadband portion of the current frame again assuming it contains voice signals, is encoded at a data rate of 14 kbit / s using the TDBWE (Time Domain Bandwidth Extension) method.
- TDBWE Time Domain Bandwidth Extension
- the speech signal is also split into a narrowband and a broadband component, with both components sampled at a frequency of 8 kHz.
- the decomposition also takes place via a QMF filter.
- the narrowband portion is encoded using narrow band SID information.
- This narrowband SID information is sent to the receiver at a later time in a SID frame compatible with the G.729 standard. Further measures as described above can contribute to an improvement of the narrowband SID component.
- the broadband component is encoded using a modified TDBWE method.
- the voice signal is further encoded at a data rate of 14 kbit / s, while at the same time the background noise detected during the speech pause is evaluated and corresponding parameters are set.
- the background noise is evaluated with regard to the energy of the noise signal and its frequency distribution.
- the temporal fine structure is not evaluated, but merely an average of the energy is formed over the frame.
- FIG. 1 shows a SID frame with separate areas for a narrowband first portion LB ("Low Band”), a broadband second portion HB ("High Band”) and an itermediary third portion ELB ("Enhanced Low Band”).
- LB narrowband first portion
- HB broadband second portion
- ELB Enhanced Low Band
- the first component LB contains encoded background noise parameters, which are encoded at a data rate of 8 kbit / s or below.
- the data length of the first component LB is, for example, 15 bits.
- the second component HB contains encoded background noise parameters, which are encoded with a data rate between 14 kbit / s and 32 kbit / s.
- the data length of the second component HB is for example 19 bits.
- the third component ELB contains encoded background noise parameters, which are encoded with a data rate of greater than 8 kbit / s, for example 12 kbit / s.
- the data length of the third component ELB is 9 bits, for example.
- Proportion ELB consists of a possibility to reproduce a noise signal in an increased quality compared to the conventional narrow-band coding method while still conforming to the standard G.729. B to stay.
- characteristics of the background noise are learned on the part of the encoder.
- the characteristics include in particular the temporal distribution as well as the spectral form of the background noise.
- a filtering method is used, which takes into account temporal and spectral parameters of the background noise from previous frames. If there are significant changes in the character or strength of the background noise, a decision based on threshold values is made as to whether there is a need to update the learned parameters.
- the embodiments relate to further details on the inclusion of the DTX method in wideband codecs such as e.g.
- the following procedure is provided. Produce a narrowband SID information to produce a G.729 or G.729.
- B-compatible SID frame first portion LB of the SID frame according to the invention
- the background noise is analyzed or "learned" during a phase which precedes a transmission of the first SID frames, with regard to the energy and / or frequency distribution.
- SID frames are sent when a significant change in the broadband portion of the background noise is detected or when an update of the narrowband SID information is to be sent.
- An implementation of this exemplary embodiment takes place in the following phases:
- a VAD procedure is used to define an active speech phase or pause.
- a hangover period is started.
- the data rate of the encoder is reduced to 14kbit / s if the previous data rate has a higher value.
- the data rate is reduced to a value of 8 kbit / s.
- the background noise for the narrowband portion is learned in an analogous manner to the procedure in standard G.729, but using a higher number of frames.
- a filtering method can be optionally applied by which it is achieved that the current frame is assigned a higher importance than the previous frame.
- the background noise is also learned in the broadband component.
- a modified TDBWE method is used to simplify the implementation, in particular to reduce the storage space requirement, which is characterized by a simplified encoding in the time domain.
- a further simplification in the modified TDBWE method can be achieved in that the encoding in the time domain only corresponds to the energy of the signal in the time domain.
- Another optional simplified encoding is to use spectral smoothing techniques because the energy in the time domain and in the frequency domain yields equal values as a result of the parsevalt theorem.
- optionally further filtering measures can be applied which have the goal of assigning a higher importance to current frames than previous frames.
- comfort noise is synthesized on the side of the decoder or receiver on the basis of the received SID frames. Changes in background noise are detected in the narrowband portion of the SID frame, with a similar procedure to G.729 being followed, although different parameters are taken into account. In the broadband component, filtered energy parameters are used to describe the background noise.
- filtered energy parameters are derived from the TDBWE parameters defined in G.729.1 using suitable low-pass filters:
- Changes in the broadband component of the energy parameters are monitored and detected by comparing the filtered energy parameters of the current noise signal with two sets of comparison values of these parameters, a set of comparison values representing the parameters. meter from the previous frame with the index idx-1.
- temp_d 20 • log (2) tenv_f ⁇ ck -tenv_f ⁇ dxA log (l ⁇ )
- NPC ⁇ 1 NB subbands spec d 20. gK 'Y ⁇ FENV FJI] - FENV L x, [i] ⁇
- temp _ ch 20 • g • ⁇ tenv _f ⁇ dx - tenv _ f last tx log (l ⁇ )
- a regular decoding mode sets in as in G.729.1.
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- Engineering & Computer Science (AREA)
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- Health & Medical Sciences (AREA)
- Signal Processing (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Computational Linguistics (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- Quality & Reliability (AREA)
- Spectroscopy & Molecular Physics (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
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Abstract
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Priority Applications (6)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
KR1020127019596A KR101364983B1 (ko) | 2008-02-19 | 2009-02-02 | Sid 프레임을 인코딩하기 위한 방법 |
CN2009801057752A CN101952886B (zh) | 2008-02-19 | 2009-02-02 | 用于对背景噪声信息进行编码的方法和装置 |
JP2010547137A JP5361909B2 (ja) | 2008-02-19 | 2009-02-02 | 背景ノイズ情報を符号化する方法および手段 |
EP09711908.5A EP2245621B1 (de) | 2008-02-19 | 2009-02-02 | Verfahren und mittel zur enkodierung von hintergrundrauschinformationen |
US12/867,969 US20100318352A1 (en) | 2008-02-19 | 2009-02-02 | Method and means for encoding background noise information |
US14/880,490 US20160035360A1 (en) | 2008-02-19 | 2015-10-12 | Method and Means of Encoding Background Noise Information |
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE102008009719.5 | 2008-02-19 | ||
DE102008009719A DE102008009719A1 (de) | 2008-02-19 | 2008-02-19 | Verfahren und Mittel zur Enkodierung von Hintergrundrauschinformationen |
Related Child Applications (2)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
US12/867,969 A-371-Of-International US20100318352A1 (en) | 2008-02-19 | 2009-02-02 | Method and means for encoding background noise information |
US14/880,490 Continuation US20160035360A1 (en) | 2008-02-19 | 2015-10-12 | Method and Means of Encoding Background Noise Information |
Publications (1)
Publication Number | Publication Date |
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WO2009103608A1 true WO2009103608A1 (de) | 2009-08-27 |
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ID=40652248
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
PCT/EP2009/051118 WO2009103608A1 (de) | 2008-02-19 | 2009-02-02 | Verfahren und mittel zur enkodierung von hintergrundrauschinformationen |
Country Status (8)
Country | Link |
---|---|
US (2) | US20100318352A1 (de) |
EP (1) | EP2245621B1 (de) |
JP (1) | JP5361909B2 (de) |
KR (2) | KR101364983B1 (de) |
CN (1) | CN101952886B (de) |
DE (1) | DE102008009719A1 (de) |
RU (1) | RU2461080C2 (de) |
WO (1) | WO2009103608A1 (de) |
Cited By (1)
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US10089993B2 (en) | 2014-07-28 | 2018-10-02 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for comfort noise generation mode selection |
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CN101483495B (zh) * | 2008-03-20 | 2012-02-15 | 华为技术有限公司 | 一种背景噪声生成方法以及噪声处理装置 |
CN103187065B (zh) | 2011-12-30 | 2015-12-16 | 华为技术有限公司 | 音频数据的处理方法、装置和*** |
EP2936486B1 (de) | 2012-12-21 | 2018-07-18 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Hinzufügung angenehmen rauschens zur modellierung eines hintergrundrauschens bei niedrigen bitraten |
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RU2696466C2 (ru) * | 2014-07-28 | 2019-08-01 | Фраунхофер-Гезелльшафт Цур Фердерунг Дер Ангевандтен Форшунг Е.Ф. | Устройство и способ для выбора режима генерирования комфортного шума |
US11250864B2 (en) | 2014-07-28 | 2022-02-15 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for comfort noise generation mode selection |
US12009000B2 (en) | 2014-07-28 | 2024-06-11 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for comfort noise generation mode selection |
Also Published As
Publication number | Publication date |
---|---|
US20100318352A1 (en) | 2010-12-16 |
DE102008009719A1 (de) | 2009-08-20 |
KR101364983B1 (ko) | 2014-02-20 |
RU2010138563A (ru) | 2012-04-10 |
US20160035360A1 (en) | 2016-02-04 |
JP2011512563A (ja) | 2011-04-21 |
JP5361909B2 (ja) | 2013-12-04 |
RU2461080C2 (ru) | 2012-09-10 |
CN101952886B (zh) | 2013-03-06 |
KR20100120217A (ko) | 2010-11-12 |
EP2245621B1 (de) | 2019-05-01 |
EP2245621A1 (de) | 2010-11-03 |
CN101952886A (zh) | 2011-01-19 |
KR20120089378A (ko) | 2012-08-09 |
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