EP2245621B1 - Verfahren und mittel zur enkodierung von hintergrundrauschinformationen - Google Patents

Verfahren und mittel zur enkodierung von hintergrundrauschinformationen Download PDF

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Publication number
EP2245621B1
EP2245621B1 EP09711908.5A EP09711908A EP2245621B1 EP 2245621 B1 EP2245621 B1 EP 2245621B1 EP 09711908 A EP09711908 A EP 09711908A EP 2245621 B1 EP2245621 B1 EP 2245621B1
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Prior art keywords
sid
background noise
narrowband
speech
parameters
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German (de)
English (en)
French (fr)
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EP2245621A1 (de
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Herve Taddei
Stefan Schandl
Panji Setiawan
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Unify GmbH and Co KG
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Unify GmbH and Co KG
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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
    • 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
    • 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
    • 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 narrowband speech signal with a data rate of 8 kbit / s is provided.
  • broadband speech codecs Wide Band Speech Codec
  • Such an extended frequency range is e.g. between a frequency of 50 Hz and 7000 Hz.
  • a wideband voice codec is known from ITU-T Recommendation G.729.EV.
  • coding methods for broadband speech codecs are made scalable.
  • scalability it is meant here that the transmitted coded data includes various demarcated blocks containing the narrowband portion, the wideband portion and / or the full bandwidth of the coded voice signal.
  • such a scalable design allows for backward 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 in the transmission channel.
  • a compression of the data to be transmitted For a reduction of the data transmission rate by a codec is usually provided a compression of the data to be transmitted. Compression is achieved, for example, by coding methods, parameters for an excitation signal and filter parameters being determined for encoding the speech data. The filter parameters and parameters specifying the excitation signal are then transmitted to the receiver. There, a synthetic speech signal is synthesized using the codec, which is as similar as possible to the original speech signal in terms of a subjective hearing impression. With the help of this method, also known as "analysis-by-synthesis", the determined and digitized samples are not transmitted, but determined parameters that allow 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 familiar in the art under the term DTX.
  • DTX discontinuous Transmission
  • the basic goal of DTX is to reduce the data transfer rate in the event of a speech break.
  • a pause detection (Voice Activity Detection, VAD) is used on the part of the transmitter, which recognizes when a certain signal level falls below a speech break.
  • VAD Voice Activity Detection
  • the receiver is not expected to have complete silence during a speech break.
  • a complete silence on the receiver side would lead to irritation or even to the suspicion 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).
  • G.729.1 SID has an embedded structure with a core SID equal to the G.729 SID and a first and second extension layer.
  • the first enhancement layer adds some parameters for narrowband comfort noise, while the second enhancement layer adds wideband information, with the SID much smaller than any other frame.
  • a marker (M) bit should be set to 1 when using DTX in the RTP header.
  • ITU-T standard G.729.1 (05/2006): SERIES G: TRANSMISSION SYSTEMS AND MEDIA, DIGITAL SYSTEMS AND NETWORKS, Digital terminal equipments - Coding of analogue signals by methods other than PCM, "G.729-based embedded variable bit-rate coder: to 8-32 kbps scalable wideband coder bitstream interoperable with G.729", ITU-T RECOMMENDATION G.729.1, approved on 29 May 2006 (2006-05 -29) by ITU-T Study Group 16 (2005-2008), International Telecommunication Union, Geneva, CH, ITU-T Rec.
  • the encoding of the background noise information occurs over either 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.
  • the reconstructed, ie synthesized, comfort noise may therefore have a different quality than the speech information synthesized on the receiver side. This adversely affects the reception of the recipient.
  • 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 encoding method provides for encoding a narrowband first and a wideband 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 second data area separate from it 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 comfort noise should be based on the broadband portion of the transmitted SID frames or on the basis of the narrowband portion.
  • This is of particular advantage for the receiver-side acoustic reception in a situation where 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 narrowband data (extended narrowband data or "enhanced low band").
  • the advantage of defining the SID frame with this third component is the ability to reproduce a noise signal in a quality enhanced in comparison to conventional narrowband coding while remaining in compliance with the G.729.B standard.
  • the single FIGURE shows a structure of a SID frame according to the invention.
  • discontinuous transmission (DTX) methods for the transmission of background noise information currently do not support the scalable character intended for the transmission of the speech information.
  • narrow-band speech codecs such as e.g. 3GPP AMR, ITU-T G.729 and on the other hand broadband speech codecs, e.g. 3GPP AMR-WB, ITU-T G.722.
  • a narrow-band 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 techniques, that is, discontinuous transmission techniques to reduce the overall transmission rate in the communication channel.
  • DTX discontinuous transmission techniques to reduce the overall transmission rate in the communication channel.
  • SID frames are transmitted, with the bandwidth of the SID frames corresponding to the bandwidth of the voice signal.
  • the background noise during a speech break is described.
  • This codec 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).
  • 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 speech 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 shows a SID frame with separate areas for a narrowband first portion LB ("Low Band”), a broadband second portion HB ("High Band”) and an itermediate 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.
  • 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 background noise from previous frames. If there are significant changes in the character or magnitude of background noise, a decision is made based on threshold values as to whether there is a need to update the learned parameters.
  • the following procedure is carried out:
  • a "regular" ie a speech signal containing, frame is received
  • the data rate for such regular frames is usually 8 kbit / s or above.
  • comfort noise is synthesized, and in the case of a wideband SID, a broadband comfort noise is synthesized and output with a read-out gain.
  • the embodiments relate to further details for incorporating the DTX method into broadband codecs such as e.g. G.729.1, and further methods for modifying the TDBWE method, which include synthesizing comfort noise during non-active frames, i. Frames without language information, support.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • 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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EP09711908.5A 2008-02-19 2009-02-02 Verfahren und mittel zur enkodierung von hintergrundrauschinformationen Active EP2245621B1 (de)

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DE102008009719A DE102008009719A1 (de) 2008-02-19 2008-02-19 Verfahren und Mittel zur Enkodierung von Hintergrundrauschinformationen
PCT/EP2009/051118 WO2009103608A1 (de) 2008-02-19 2009-02-02 Verfahren und mittel zur enkodierung von hintergrundrauschinformationen

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EP (1) EP2245621B1 (ja)
JP (1) JP5361909B2 (ja)
KR (2) KR101364983B1 (ja)
CN (1) CN101952886B (ja)
DE (1) DE102008009719A1 (ja)
RU (1) RU2461080C2 (ja)
WO (1) WO2009103608A1 (ja)

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CN101952886A (zh) 2011-01-19
EP2245621A1 (de) 2010-11-03
KR101364983B1 (ko) 2014-02-20
WO2009103608A1 (de) 2009-08-27
KR20100120217A (ko) 2010-11-12
JP5361909B2 (ja) 2013-12-04
JP2011512563A (ja) 2011-04-21
KR20120089378A (ko) 2012-08-09
RU2461080C2 (ru) 2012-09-10
CN101952886B (zh) 2013-03-06
US20160035360A1 (en) 2016-02-04
RU2010138563A (ru) 2012-04-10
DE102008009719A1 (de) 2009-08-20
US20100318352A1 (en) 2010-12-16

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