EP2380169B1 - Method and means for the scalable improvement of the quality of a signal encoding method - Google Patents
Method and means for the scalable improvement of the quality of a signal encoding method Download PDFInfo
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- EP2380169B1 EP2380169B1 EP09807441.2A EP09807441A EP2380169B1 EP 2380169 B1 EP2380169 B1 EP 2380169B1 EP 09807441 A EP09807441 A EP 09807441A EP 2380169 B1 EP2380169 B1 EP 2380169B1
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- 238000000034 method Methods 0.000 title claims description 25
- 230000005236 sound signal Effects 0.000 claims description 6
- 230000003044 adaptive effect Effects 0.000 claims description 5
- 238000004393 prognosis Methods 0.000 claims description 3
- 238000012804 iterative process Methods 0.000 claims description 2
- 230000005540 biological transmission Effects 0.000 description 6
- 238000005070 sampling Methods 0.000 description 5
- VKZRWSNIWNFCIQ-WDSKDSINSA-N (2s)-2-[2-[[(1s)-1,2-dicarboxyethyl]amino]ethylamino]butanedioic acid Chemical compound OC(=O)C[C@@H](C(O)=O)NCCN[C@H](C(O)=O)CC(O)=O VKZRWSNIWNFCIQ-WDSKDSINSA-N 0.000 description 1
- 101150072497 EDS1 gene Proteins 0.000 description 1
- 230000006978 adaptation Effects 0.000 description 1
- 238000004891 communication Methods 0.000 description 1
- 238000010276 construction Methods 0.000 description 1
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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 a method for the scalable improvement of the quality of a signal coding method.
- the audio signals to be transmitted are compressed by means of coding methods and decompressed after the transmission.
- Such a coding method which is intended for transmission of a speech signal in a frequency range of 300 to 3400 Hz with a data rate of 8 kbit / s, is known, for example, from ITU-T Recommendation G.729.
- broadband voice codec For transmission of a higher quality, the transmission of an extended frequency range of 50 Hz to 7000 Hz is known.
- a designated, so-called broadband voice codec is described, for example, in ITU-T Recommendation G.722.EV.
- the method uses the so-called Subband-Adaptive Differential Pulse Code Modulation (SB-ADPCM) for encoding audio signals, which in the ITU-T Recommendation G.722, "7 kHz Audio Coding within 64 kbit / s"; G.722 (11/88), International Telecommunication Union, Geneva, CH, November 25, 1988 , XP017400870.
- SB-ADPCM Subband-Adaptive Differential Pulse Code Modulation
- the invention has for its object to provide a method for scalable improvement of the quality of a coding method according to the Subband Adaptive Differential Pulse Code principle.
- the object is achieved by claim 1 with a method for scalable improvement of the quality of a coding method according to IT-U Recommendation G.722 with the following method steps,
- M-1 indicates the distance of the first pulse from the beginning of the comparison section, ⁇ p ⁇ ⁇ 0 , ⁇ 1 , ⁇ , ⁇ L -1 ⁇ the amplitude value, M den Distance between two individual Pulses, N the number of pulses and L the number of different levels ⁇ ; The information about the reference signal with the minimum error signal is transmitted.
- the reference signal according to Fig. 1 comprises a number of N Dirac pulses ⁇ ( n ).
- the Dirac pulses may have a predetermined number of amplitude values L.
- Fig. 2 shows the structural structure of an encoder according to the invention, which in addition to a conventional encoder ADPCM according to the subband Adaptive Differential Pulse Code principle according to IT-U Recommendation G.722 means for generating reference signals comprising for each stage of the extension in each case a signal generator EHDS1, .. EHDSS for generating the reference signals c (n) and a control unit CB 1, ... CB S has.
- the reference signals c (n) are compared over a given time interval, a so-called frame, with a digital error signal e H , which in a conventional coding method according to IT-U Recommendation G.722 was determined from an input signal to be coded and a prognosis signal.
- the reference signal c (n) with the lowest error amount E n is now determined and the information about this signal is transmitted as additional information I H1min ,... I HSmin and used in the receiver for decoding the useful signal ,
- the starting point is a sampling rate of 8 KHz and thus a duration of a sampling interval of 125 ⁇ sec.
- the duration of a comparison section is 5 msec, the number of possible amplitude values L for the Dirac pulses is 2.
- Fig. 3 shows the structure of a decoder according to the invention in which the audio signal is obtained from the received signal I H , I H1 , I H2 ... I HS .
- the received signal comprises, in addition to the output signal I H of the conventional coder ADPCM, the additional information I H1min ,... I HSmin obtained with the invention as a function of the number of expansion stages selected in the transmitter.
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- Engineering & Computer Science (AREA)
- Quality & Reliability (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)
Description
Die Erfindung betrifft ein Verfahren zur skalierbaren Verbesserung der Qualität eines Signalcodierverfahrens In digitalen Kommunikationssystemen werden zur Reduktion der benötigen Datenraten die zu übertragenden Audiosignale mittels Kodierungsverfahren komprimiert und nach der Übertragung dekomprimiert.The invention relates to a method for the scalable improvement of the quality of a signal coding method. In digital communication systems, to reduce the required data rates, the audio signals to be transmitted are compressed by means of coding methods and decompressed after the transmission.
Ein derartiges Codierverfahren, welches für eine Übertragung eines Sprachsignals in einem Frequenzbereich von 300 bis 3400 Hz mit einer Datenrate von 8 kbit/s vorgesehen ist, ist beispielsweise aus der ITU-T-Empfehlung G.729 bekannt.Such a coding method, which is intended for transmission of a speech signal in a frequency range of 300 to 3400 Hz with a data rate of 8 kbit / s, is known, for example, from ITU-T Recommendation G.729.
Für die Übertragung mit einer höheren Qualität ist die Übertragung eines erweiterten Frequenzbereiches von 50 Hz bis 7000 Hz bekannt. Ein hierfür vorgesehener, sogenannter breitbandiger Sprach-Codec ist beispielsweise in der ITU-T-Empfehlung G.722.EV beschrieben.For transmission of a higher quality, the transmission of an extended frequency range of 50 Hz to 7000 Hz is known. A designated, so-called broadband voice codec is described, for example, in ITU-T Recommendation G.722.EV.
Das Verfahren nutzt die sogenannte Subband- Adaptive Differential Pulse Code Modulation (SB-ADPCM) zur Codierung von Audiosignalen, das in der
Zur weiteren Erhöhung der Qualität des übertragenen Audiosignales besteht die Forderung nach einem skalierbaren Codierungsverfahren.To further increase the quality of the transmitted audio signal, there is a demand for a scalable coding method.
Die Skalierbarkeit gestattet einerseits eine empfängerseitige Abwärtskompatibilität mit herkömmlichen Dekodierungsverfahren und andererseits bietet sie eine einfache Möglichkeit, im Falle von eingeschränkten Datenübertragungskapazitäten im Übertragungskanal eine sender- und empfängerseitige Anpassung der Datenrate und der Größe von übertragenen Datenrahmen vorzunehmen.On the one hand, scalability allows backward compatibility with conventional decoding methods and on the other hand, it offers a simple possibility, in the case of limited data transmission capacities in the transmission channel, to carry out a transmitter and receiver-side adaptation of the data rate and the size of transmitted data frames.
Der Erfindung liegt die Aufgabe zugrunde, ein Verfahren zur skalierbaren Verbesserung der Qualität eines Codierungsverfahrens nach dem Subband- Adaptive Differential Pulse Code Prinzip anzugeben.The invention has for its object to provide a method for scalable improvement of the quality of a coding method according to the Subband Adaptive Differential Pulse Code principle.
Die Aufgabe wird erfindungsgemäß durch Anspruch 1 gelöst mit einem Verfahren zur skalierbaren Verbesserung der Qualität eines Codierungsverfahrens gemäß IT-U Recommendation G.722 mit folgenden Verfahrensschritten,
in einem iterativen Prozess mit einer vom Umfang der Erweiterung abhängigen Anzahl von Wiederholungsschritten wird ein aus einem zu codierenden Eingangssignal und einem Prognosesignal ermitteltes digitales Fehlersignal abschnittsweise mit einer Anzahl von M*LN unterschiedlichen Referenzsignalen verglichen und daraus das Referenzsignal mit einem minimalen Fehlersignal hinsichtlich eines vorgegebenen Fehlerkriteriums ermittelt,
die Referenzsignale c(n) sind jeweils aus äquidistanten Dirac-Impulsen δ(n) gemäß
Die Information über das Referenzsignal mit dem minimalen Fehlersignal wird übertragen.The object is achieved by
In an iterative process with a number of repetition steps depending on the extent of the extension, a digital error signal determined from an input signal to be coded and a prognosis signal is compared in sections with a number of M * L N different reference signals, and from this the reference signal with a minimum error signal with respect to a predetermined one Error criterion determined,
the reference signals c (n) are each made of equidistant Dirac pulses δ ( n ) according to
The information about the reference signal with the minimum error signal is transmitted.
Dabei ist es vorteilhaft, wenn als Fehlerkriterium ein erweitertes Fehlersignal eH1(n) gemäß
Vorteilhaft ist auch eine Anordnung zur Durchführung des erfindungsgemäßen Verfahrens nach Anspruch 3, bei dem neben einem herkömmlichen Kodierer (ADPCM) nach dem Subband Adaptive Differential Pulse Code Prinzip gemäß IT-U Recommendation G.722 Mittel zur Erstellung von Referenzsignalen vorgesehen sind, die für jede Stufe der Erweiterung jeweils einen Signalgenerator EHDS1 ,... EHDSS zur Erzeugung der Referenzsignale c(n) und eine Steuereinheit CB 1, ...CB S aufweisen.Also advantageous is an arrangement for carrying out the inventive method according to
Im Folgenden wird eine Ausführungsform des erfindungsgemäßen Verfahrens anhand der Figuren erläutert.In the following, an embodiment of the method according to the invention will be explained with reference to the figures.
Es zeigen beispielhaft:
-
Fig.1 den Aufbau eines erfindungsgemäßen Referenzsignales -
Fig. 2 die Struktur eines erfindungsgemäßen Codecs und -
Fig. 3 die Struktur eines erfindungsgemäßen Decoders.
-
Fig.1 the structure of a reference signal according to the invention -
Fig. 2 the structure of a codec according to the invention and -
Fig. 3 the structure of a decoder according to the invention.
Das Referenzsignal gemäß
Die mathematische Definition eines Referenzsignales ist wie folgt:
Durch Variation der Parameter Amplitudenwert α mit L unterschiedlichen Werten und des Offsets off = [0 .. M-1]wird nun eine Gruppe mit einer Anzahl M·LN von unterschiedlichen Referenzsignalen erzeugt.By varying the parameters amplitude value α with different values of L and offset off = [0 .. M-1] will be generated by different reference signals a group having a number M · L N.
Der erfindungsgemäße Vergleich der solcherart erhaltenen Referenzsignale c(n) wird anhand der
Erfindungsgemäß werden die Referenzsignale c(n) über einen vorgegebenen zeitlichen Abschnitt,, ein sogenanntes Frame, mit einem digitalen Fehlersignal eH verglichen, welches in einem herkömmlichen Codierungsverfahren gemäß IT-U Recommendation G.722 aus einem zu codierenden Eingangssignal und einem Prognosesignal ermittelt wurde.According to the invention, the reference signals c (n) are compared over a given time interval, a so-called frame, with a digital error signal e H , which in a conventional coding method according to IT-U Recommendation G.722 was determined from an input signal to be coded and a prognosis signal.
Daraus ergibt sich gemäß eH1(n)= eH -c(n) ein erweitertes Fehlersignal eH1(n) zu dem über den Zeitraum des Vergleichabschnittes gemäß
Mittels Steuereinheit CB 1, ...CB S wird nun das Referenzsignal c(n) mit dem geringsten Fehlerbetrag En ermittelt undie Information über dieses Signal als zusätzliche Information IH1min,...IHSmin übertragen und im Empfänger zur Dekodierung des Nutzsignales herangezogen.By means of the
Dabei haben sich in der Praxis für den Aufbau des Referenzsignales c(n) folgende Parameter bewährt.The following parameters have proven useful in practice for the construction of the reference signal c (n).
Ausgangspunkt ist eine Abtastrate von 8 KHz und damit eine Dauer eines Abtastintervalles von 125 µsec. Die Dauer eines Vergleichsabschnittes beträgt 5 msec, die Anzahl der möglichen Amplitudenwerte L für die Dirac-Impulse beträgt 2. Die Zahl der Dirac-Impulse selbst in einem Vergleichsabschnitt beträgt N = 5. Der Abstand zwischen jeweils 2 Dirac-Impulsen beträgt M= 8 Abtastintervalle.The starting point is a sampling rate of 8 KHz and thus a duration of a sampling interval of 125 μsec. The duration of a comparison section is 5 msec, the number of possible amplitude values L for the Dirac pulses is 2. The number of Dirac pulses itself in a comparison section is N = 5. The distance between every 2 Dirac pulses is M = 8 sampling intervals ,
Der oben beschriebene Vergleichsvorgang der Referenzsignale c(n) mit dem digitalen Fehlersignal eH wird nun in Abhängigkeit von der gewählten Skalierung iterativ mehrfach wiederholt, was in der
Das heißt für den ersten Wiederholungsschritt, dass die Referenzsignale c(n) mit dem erweiterten ersten Fehlersignal eH1(n) verglichen und daraus ein erweitertes zweites Fehlersignal EH2(n) generiert wird. Dieser Vorgang wird typisch viermal wiederholt.For the first repetition step, this means that the reference signals c (n) are compared with the extended first error signal e H1 (n) and an extended second error signal E H2 (n) is generated therefrom. This process is typically repeated four times.
Dabei besteht ein wesentlicher Vorteil auch darin, dass nicht alle Informationen, die im empfangenen Signal vorhanden sind, auch tatsächlich ausgewertet werden müssen. So is es möglich, dass ein Empfänger mit nur einem herkömmlichen Dekoder Core Decoder ein Signal empfängt, welches auch die Zusatzinformationen IH1min,... IHSmin enthält, diese aber für die Gewinnung des Audiosignales nicht heranziehtIn this case, a significant advantage is also that not all information that is present in the received signal, actually need to be evaluated. It is thus possible that a receiver with only one conventional decoder core decoder receives a signal which also contains the additional information I H1min ,... I HSmin , but does not use it for the extraction of the audio signal
Diese Möglichkeit wird als Abwärtskompatibilität bezeichnet.This possibility is called backward compatibility.
Bei einem Empfänger, der die erfindungsgemäßen Erweiterungsstufen EDS1, EDS2, ... EDSS zur Dekodierung der Zusatzinformationen IH1min,...IHSmin umfasst, wird hingegen das Signal in voller Qualität dekodiert, sofern nicht aus anderen Gründen eine Einschränkung geboten ist.In the case of a receiver which includes the inventive expansion stages EDS1, EDS2,... EDSS for decoding the additional information I H1min ,... I HSmin , the signal is decoded in full quality, unless a restriction is required for other reasons.
Claims (3)
- A method for scalable improvement of the quality of an encoding method, in which the Subband Adaptive Differential Pulse Code Modulation (SB-ADPCM) for encoding audio signals pursuant to IT-U Recommendation G.722 is used, comprising the following method steps,- comparing a digital error signal (eH), derived from an input signal to be encoded and a prognosis signal, in sections to a number of M*LN different reference signals c(n) in an iterative process having a number of repeated steps depending on a scope of an expansion, and deriving from said comparison a reference signal having a minimum error signal with respect to a prescribed error criterion,- each of the reference signals c(n) is made up of equidistant Dirac impulses δ(n) according to the formula
α P ∈{α0,α1,···,α L-1} indicates the amplitude value, M the distance between two individual pulses, N the number of pulses, L the number of different levels α; transmitting the information (IHimin, ... IHSmin) about the reference signal with the minimum error signal. - An arrangement designed for carrying out the method according to Claims 1 or 2, wherein in addition to a conventional encoder (ADPCM) operating according to the Subband Adaptive Differential Pulse Code principle pursuant to IT-U Recommendation G.722, means are provided for generating reference signals which, for each step of the expansion, have a signal generator (EHDS 1 ,... EHDSS) for generating the reference signals c(n), and a control unit (CB 1, ... CB S).
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
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ATA1982/2008A AT509439B1 (en) | 2008-12-19 | 2008-12-19 | METHOD AND MEANS FOR SCALABLE IMPROVEMENT OF THE QUALITY OF A SIGNAL CODING METHOD |
PCT/EP2009/008853 WO2010069513A1 (en) | 2008-12-19 | 2009-12-10 | Method and means for the scalable improvement of the quality of a signal encoding method |
Publications (2)
Publication Number | Publication Date |
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EP2380169A1 EP2380169A1 (en) | 2011-10-26 |
EP2380169B1 true EP2380169B1 (en) | 2015-12-09 |
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EP09807441.2A Active EP2380169B1 (en) | 2008-12-19 | 2009-12-10 | Method and means for the scalable improvement of the quality of a signal encoding method |
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US (1) | US8774312B2 (en) |
EP (1) | EP2380169B1 (en) |
CN (1) | CN102257565B (en) |
AT (1) | AT509439B1 (en) |
BR (1) | BRPI0922993A2 (en) |
WO (1) | WO2010069513A1 (en) |
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FR2481026B1 (en) * | 1980-04-21 | 1984-06-15 | France Etat | |
JP2598159B2 (en) * | 1990-08-28 | 1997-04-09 | 三菱電機株式会社 | Audio signal processing device |
US5956674A (en) * | 1995-12-01 | 1999-09-21 | Digital Theater Systems, Inc. | Multi-channel predictive subband audio coder using psychoacoustic adaptive bit allocation in frequency, time and over the multiple channels |
WO2003077235A1 (en) * | 2002-03-12 | 2003-09-18 | Nokia Corporation | Efficient improvements in scalable audio coding |
KR100467326B1 (en) * | 2002-12-09 | 2005-01-24 | 학교법인연세대학교 | Transmitter and receiver having for speech coding and decoding using additional bit allocation method |
-
2008
- 2008-12-19 AT ATA1982/2008A patent/AT509439B1/en not_active IP Right Cessation
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2009
- 2009-12-10 CN CN2009801510367A patent/CN102257565B/en not_active Expired - Fee Related
- 2009-12-10 US US13/133,978 patent/US8774312B2/en active Active
- 2009-12-10 BR BRPI0922993A patent/BRPI0922993A2/en not_active Application Discontinuation
- 2009-12-10 EP EP09807441.2A patent/EP2380169B1/en active Active
- 2009-12-10 WO PCT/EP2009/008853 patent/WO2010069513A1/en active Application Filing
Also Published As
Publication number | Publication date |
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AT509439B1 (en) | 2013-05-15 |
WO2010069513A1 (en) | 2010-06-24 |
CN102257565A (en) | 2011-11-23 |
AT509439A1 (en) | 2011-08-15 |
EP2380169A1 (en) | 2011-10-26 |
CN102257565B (en) | 2013-05-29 |
BRPI0922993A2 (en) | 2016-01-26 |
US8774312B2 (en) | 2014-07-08 |
US20120014474A1 (en) | 2012-01-19 |
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