EP2245622B1 - Procédés et moyens pour décoder des informations de bruit de fond - Google Patents

Procédés et moyens pour décoder des informations de bruit de fond Download PDF

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Publication number
EP2245622B1
EP2245622B1 EP09712583.5A EP09712583A EP2245622B1 EP 2245622 B1 EP2245622 B1 EP 2245622B1 EP 09712583 A EP09712583 A EP 09712583A EP 2245622 B1 EP2245622 B1 EP 2245622B1
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Prior art keywords
time
entrance
broadband
phase
dtx
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German (de)
English (en)
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EP2245622A1 (fr
Inventor
Panji Setiawan
Stefan Schandl
Herve Taddei
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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
    • 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
    • G10L19/04—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 decoding 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 Bandeumbleabegrezung the analog signal Prior to a coding process, a Bandeumbleabegrezung 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 lies for example between a frequency of 50 Hz and 7000 Hz.
  • a broadband speech codec is known, for example, from the ITU-T Recommendation G.729.EV.
  • coding methods for broadband speech codecs are made scalable.
  • a scalability it is meant that the transmitted encoded data exhibits various demarcated blocks containing the narrowband portion, wideband portion and / or full bandwidth of the encoded 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, in which parameters for an excitation signal and filter parameters are determined for coding 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 referred to as "analysis-by-synthesis", the determined and digitized samples (samples) are not transmitted themselves, but determined parameters which 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 familiar in the art under the term DTX.
  • the fundamental goal DTX is a reduction in the data transfer rate in the event of a speech break.
  • a voice pause detection (VAD) is used on the part of the transmitter, which detects when a certain signal level falls below a speech break.
  • 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 synthesis - i.
  • Decoding - comfort noise transmits data at a much lower data rate.
  • the data transmitted here are also referred to in the art as SID (Silence Insertion Description).
  • Said scalable wideband speech codecs usually support different data transmission rates in a bandwidth range of 50 to 7000 Hz.
  • Possible data rates for encoding the speech information are e.g. 8, 12, 14, 16, ..., 32 kbit / s, which is e.g. are used in standard G.729.1.
  • the data rates of 8 and 12 kbit / s are applied to narrowband signals (50Hz to 4kHz).
  • Data rates greater than 12 kbit / s are applied to the upper frequency band of 4 to 7 kHz.
  • a sudden change from a narrowband to a broadband data rate is known to cause a disturbing effect on a human recipient.
  • Such a transition occurs, for example, as a result of a truncation of the data stream (bitstream transduction), which is caused, for example, by the transmission network between transmitter and receiver, for example as a result of the establishment of further additional connections or due to a congestion in the transmission network.
  • bitstream transduction causes, for example, by the transmission network between transmitter and receiver, for example as a result of the establishment of further additional connections or due to a congestion in the transmission network.
  • the said truncation leads to a change in the data rate and finally to a transition from a broadband to a narrowband transmission of the speech signal.
  • a saving of the data transmission rate for the transmission of the respective data frames is possible.
  • the DTX method is used exactly when a corresponding frame is characterized as a speech break.
  • a reduced data transfer rate on transmitted frames is achieved due to two factors. First, the encoder does not need to send all inactive frames to the decoder. Second, a transmitted SID frame occupies much fewer bits than a voice data frame.
  • Such a method requires involvement of a speech pause recognition (VAD) on the encoder side.
  • VAD speech pause recognition
  • the encoder on the transmitter side is informed whether a current sample containing and to encoded frame contains a speech signal or a speech break with background noise.
  • this characterization measures are taken in the encoder which determine the perceptional characteristics of an inactive frame (Inactive Speech Frame).
  • perception characteristics include, for example, the average energy and spectral and temporal characteristics.
  • the encoder then sends a specially marked frame, a Silence Insertion Descriptor (SID) frame, to the decoder.
  • SID Silence Insertion Descriptor
  • the decoder synthesizes comfort noise based on the information contained in a SID frame, and the decoder can determine, based on the SID frame, whether the contained noise information is narrowband or broadband information.
  • bit rate switching Changing the bit rate (bit rate switching") between narrowband and wideband information is a common scenario for any scalable wideband speech codec.
  • the object of the invention is to specify a method for changing a data rate of SID frames during a speech pause, which results in an improved quality of the signal synthesized on the decoder.
  • a basic idea of the invention is to carry out a determination of information about the course of the bit rate switching during an active speech phase.
  • the scalable nature of the speech signal coding method or codec used according to the invention expresses while the possibility of the codec for bandwidth switching already off.
  • the decoder during the speech phase on the part of the decoder, information is collected on the percentage of broadband active speech frames in comparison to narrow-band active speech frames.
  • information on the nature of the background noise is not collected until the time of a change to a speech pause, as has hitherto been suggested by the prior art.
  • a higher percentage of broadband active speech frames indicates that a broadband use is preferred by the codec and therefore there is a need to broadly synthesize noise information during a DTX phase, i. to decode.
  • narrow-band noise is generated on the decoder's side when entering a DTX phase, even if the received SID frames would allow for synthesizing - i.e. decoding - a wideband noise.
  • the object of the invention to provide a method for changing a data rate of SID frames during a speech break more than solved.
  • the change to be made between noise information with different data rate according to the object is refined according to the inventive solution presented here into a determination of a proportion of noise information with different data rates.
  • the proportion is adjustable in contrast to a change in any ratio between noise information with different data rate.
  • the adjustability or adaptability of the noise signal quality to the previously raised speech signal quality results for the entire signal, ie noise and speech signal, on the receiver side overall a significantly increased signal quality.
  • the method according to the invention thus achieves the object of the invention to achieve an improved quality of the signal synthesized on the decoder.
  • a decision is made that a noise signal having a certain quality (ie, wideband or narrowband) is synthesized during a speech pause a case may occur in which a truncation occurs in the last few frames during an active speech phase on the part of the network the active data frame took place.
  • a predominantly narrow-band decoding of the background noise information first takes place, which after a settable period of time transitions into a predominantly wideband decoding.
  • Such a transition is thus preferably quasi-continuous, with a Transition to discrete times - hence "quasi" -continuous - is set to a certain share factor.
  • the codec used was a narrow-band playback mode Favor and / or a broadband transmission in the past by the transmission network was not guaranteed. This may lead to the case that few active speech frames arrive at the receiving decoder as wideband speech frames before receiving first SID frames there.
  • a predominantly wideband decoding of the background noise information initially takes place, which after a settable period of time transitions into a predominantly narrowband decoding.
  • Such a transition is preferably quasi-continuous analogous to the development described above, wherein a transition to discrete times is set to a certain proportion factor.
  • the proportional factor is set with values as above, but in reverse order.
  • Fig. 1 is a temporal transmission of Pub Schemerah-5 men with a respective data rate DR and, as of a third time t3, a transmission of SID frames shown.
  • a transmission of broadband active speech frames with a data rate of 32 kbit / s.
  • a switchover to a data rate of 22 kbit / s and from a second time t2 to a data rate of 12 kbit / s.
  • a data rate of 12 kbit / s already corresponds to a narrowband speech frame.
  • FIG. 2A and FIG. 2B show two possible scenarios for a progression of the data rate DR over time t.
  • FIG. 2A is due to limitations of the network or due to other circumstances a transmission largely narrowband, in the example of Fig. 2A with 8. kbit / s, while at a few points in time, between a first time t1 and a second time t2, exceptionally, a broadband transmission with 32 kbit / s takes place.
  • FIG. 2B again, there is a reverse situation, namely a predominantly broadband transmission mode with 32 kbit / s and exceptionally short, between a fourth time t4 and a fifth time t5 takes place, narrow-band transmission.
  • Fig. 2A the percentage of broadband active speech frames is to be described as very low, while in the example of Fig. 2B a high percentage Proportion of broadband active speech frames.
  • Fig. 3 is a noise signal quality HB-SHARE over a time TIME, which is indicated in ms, plotted.
  • the Fig. 3 In this case, a design of the noise signal in connection with a scenario according to the preceding FIG. 2B in which, based on the decoder-side determined percentage of broadband active speech frames, a need has been identified for broadband synthesizing noise information during the DTX phase.
  • a further embodiment of the invention analogously provides for a transition from a wideband speech signal to a narrowband noise signal.
  • Fig. 2A adopted a slightly modified scenario in which, unlike in Fig. 2A represented scenario shortly before the time t3 still not shown - change to a broadband transmission at 32 kbit / s takes place. Despite this "peak", the percentage of broadband active speech frames remains very low, so that now on transition to the DTX phase, a noise signal is to be synthesized that broadband begins, however - due to the predominantly narrow-band transmission history and thus expected for the future Continuation of the narrow-band transmission character - is to be converted into a narrow-band noise signal.
  • the entry into the DTX phase is started with an exclusively broadband signal, ie with a HB-SHARE component of the broadband noise of 1.
  • the narrow-band noise component 0.
  • the values proposed above are advantageously set in an inverse row. This would correspond to a curve mirrored at the ordinate HB-SHARE Fig. 3 ,

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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)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Telephonic Communication Services (AREA)

Claims (15)

  1. Procédé de décodage d'une trame SID (SID) pour une transmission d'informations de bruit de fond en application d'un procédé de codage de signal vocal modulable avec les étapes suivantes :
    la détermination d'une proportion de trames vocales à large bande reçues par rapport à des trames vocales à bande étroite reçues pendant une phase vocale,
    le décodage des informations de bruit de fond contenues dans une trame SID lors de l'entrée dans une phase DTX, dans lequel le décodage est effectué conformément à la proportion déterminée principalement à large bande ou bande étroite.
  2. Procédé selon la revendication 1, caractérisé en ce que, dans le cas d'une détermination d'une proportion élevée de trames vocales à large bande reçues lors de l'entrée dans la phase DTX, un décodage principalement à large bande des informations de bruit de fond est effectué.
  3. Procédé selon la revendication 2, caractérisé en ce que lors de l'entrée dans la phase DTX, un décodage principalement à bande étroite des informations de bruit de fond est, tout d'abord, effectué, lequel décodage passe, après une période réglable à un décodage principalement à large bande.
  4. Procédé selon la revendication 3, caractérisé en ce que le passage au décodage principalement à large bande est réglable avec un facteur de proportion (HB-SHARE) qui exprime un rapport entre une qualité de signal de bruit à large bande et une qualité de signal de bruit à bande étroite.
  5. Procédé selon la revendication 4, caractérisé en ce que le facteur de proportion est dimensionné à zéro au moment de l'entrée dans la phase DTX.
  6. Procédé selon l'une des revendications 4 à 5, caractérisé en ce que le facteur de proportion est dimensionné à un, à un moment de 100 ms après l'entrée dans la phase DTX.
  7. Procédé selon l'une des revendications 4 à 6, caractérisé en ce que le facteur de proportion
    - est dimensionné à un moment de 20 ms après l'entrée dans la phase DTX à 0,09525986892242 ;
    - est dimensionné à un moment de 40 ms après l'entrée dans la phase DTX à 0,19753086419753 ;
    - est dimensionné à un moment de 60 ms après l'entrée dans la phase DTX à 0,36595031245237 ;
    - est dimensionné à un moment de 80 ms après l'entrée dans la phase DTX à 0,62429507696997.
  8. Procédé selon la revendication 1, caractérisé en ce que, dans le cas d'une détermination d'une proportion faible de trames vocales à large bande reçues lors de l'entrée dans la phase DTX, un décodage principalement à bande étroite des informations de bruit de fond est effectué.
  9. Procédé selon la revendication 8, caractérisé en ce que, lors de l'entrée dans la phase DTX, un décodage principalement à large bande des informations de bruit de fond est tout d'abord effectué, lequel décodage passe, après une période réglable à un décodage principalement à bande étroite.
  10. Procédé selon la revendication 9, caractérisé en ce que le passage au décodage principalement à bande étroite est réglable avec un facteur de proportion (HB-SHARE) qui exprime un rapport entre une qualité de signal de bruit à large bande et une qualité de signal de bruit à bande étroite.
  11. Procédé selon la revendication 10, caractérisé en ce que le facteur de proportion est dimensionné à un au moment de l'entrée dans la phase DTX.
  12. Procédé selon l'une des revendications 10 à 11, caractérisé en ce que le facteur de proportion est dimensionné à un moment de 100 ms après l'entrée dans la phase DTX à zéro.
  13. Procédé selon l'une des revendications 10 à 12, caractérisé en ce que le facteur de proportion
    - est dimensionné à un moment de 20 ms après l'entrée dans la phase DTX à 0,62429507696997 ;
    - est dimensionné à un moment de 40 ms après l'entrée dans la phase DTX à 0,36595031245237 ;
    - est dimensionné à un moment de 60 ms après l'entrée dans la phase DTX à 0,19753086419753 ; et ;
    - est dimensionné à un moment de 80 ms après l'entrée dans la phase DTX à 0,09525986892242.
  14. Codec avec des moyens pour la réalisation du procédé selon l'une des revendications 1 à 13.
  15. Codec selon la revendication 14, caractérisé par une implémentation dans la norme UIT-T G.729.1 connue en elle-même.
EP09712583.5A 2008-02-19 2009-02-02 Procédés et moyens pour décoder des informations de bruit de fond Active EP2245622B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE102008009720A DE102008009720A1 (de) 2008-02-19 2008-02-19 Verfahren und Mittel zur Dekodierung von Hintergrundrauschinformationen
PCT/EP2009/051120 WO2009103609A1 (fr) 2008-02-19 2009-02-02 Procédés et moyens pour décoder des informations de bruit de fond

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EP2245622A1 EP2245622A1 (fr) 2010-11-03
EP2245622B1 true EP2245622B1 (fr) 2016-07-13

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US (1) US8260606B2 (fr)
EP (1) EP2245622B1 (fr)
JP (1) JP5006975B2 (fr)
KR (1) KR101166650B1 (fr)
CN (1) CN101946281B (fr)
DE (1) DE102008009720A1 (fr)
RU (1) RU2454737C2 (fr)
WO (1) WO2009103609A1 (fr)

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JP2016038513A (ja) * 2014-08-08 2016-03-22 富士通株式会社 音声切替装置、音声切替方法及び音声切替用コンピュータプログラム
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RU2805130C1 (ru) * 2023-02-03 2023-10-11 Федеральное государственное бюджетное образовательное учреждение высшего образования "Рязанский государственный радиотехнический университет имени В.Ф. Уткина" (ФГБОУ "РГРТУ", РГРТУ) Способ построения адаптивного к акустическим шумам кодека речевых сигналов

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JP5006975B2 (ja) 2012-08-22
EP2245622A1 (fr) 2010-11-03
RU2454737C2 (ru) 2012-06-27
CN101946281B (zh) 2012-08-15
US20110040560A1 (en) 2011-02-17
JP2011512564A (ja) 2011-04-21
CN101946281A (zh) 2011-01-12
KR20100125340A (ko) 2010-11-30
WO2009103609A1 (fr) 2009-08-27
RU2010138566A (ru) 2012-03-27
DE102008009720A1 (de) 2009-08-20
US8260606B2 (en) 2012-09-04
KR101166650B1 (ko) 2012-07-23

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