EP2176861A1 - Hierarchical coding of digital audio signals - Google Patents
Hierarchical coding of digital audio signalsInfo
- Publication number
- EP2176861A1 EP2176861A1 EP08806166A EP08806166A EP2176861A1 EP 2176861 A1 EP2176861 A1 EP 2176861A1 EP 08806166 A EP08806166 A EP 08806166A EP 08806166 A EP08806166 A EP 08806166A EP 2176861 A1 EP2176861 A1 EP 2176861A1
- Authority
- EP
- European Patent Office
- Prior art keywords
- bits
- quantization
- bit
- law
- extension
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
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Classifications
-
- 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 present invention relates to a method for hierarchical coding of sound data more particularly for scalar quantization coding.
- This coding is adapted in particular for the transmission and / or storage of digital signals such as audio-frequency signals (speech, music or other).
- the present invention relates more particularly to the coding of waveforms such as the coding MIC (for "Coded Pulse Modulation") said PCM (for "Pulse Code Modulation”) in English where each input sample is individually coded, without prediction.
- waveforms such as the coding MIC (for "Coded Pulse Modulation") said PCM (for "Pulse Code Modulation”) in English where each input sample is individually coded, without prediction.
- PCM coding / decoding specified by ITU-T Recommendation G.711 is as described with reference to Figure 1.
- the input signal is assumed to be defined with a minimum bandwidth of [300-3400 Hz] and sampled at 8 kHz, with a resolution of 16 bits per sample (in so-called "linear PCM" format).
- the encoder MIC 13 comprises a quantization module Q M ic 10 which receives as input the input signal S.
- the quantization index I MIC at the output of the quantization module 10 is transmitted via the transmission channel 11 to the decoder 14.
- the decoder MIC 14 receives at the input the indices I ' MIC coming from the transmission channel, version possibly disturbed by binary errors of I MIC , and carries out an inverse quantization by the inverse quantization module Q ' Vic 12 to obtain the coded signal S ' MIC -
- the ITU-T G.711 standardized MIC coding (hereinafter G.711) compresses the signal amplitude by a logarithmic curve before uniform scalar quantization, thereby obtaining a signal-to-noise ratio near constant for a wide dynamic signals.
- the quantization step in the domain of the original signal is therefore proportional to the amplitude of the signals.
- the successive samples of the compressed signal are quantized on 8 bits, ie 256 levels.
- PSTN public switched telephone network
- PSTN Telephone Network
- these 8 bits are transmitted at a frequency of 8 kHz to give a bit rate of 64 kbit / s.
- a quantized signal frame according to the G.711 standard consists of 8-bit coded quantization indices.
- the inverse quantization is implemented per table, it simply consists of a pointing by the index of one of the 256 possible decoded values.
- the PCM compression has been approximated by a linear curve by segments.
- coding laws make it possible to perform on the signal a compression of the amplitude (or else "companding" in English).
- the amplitude of the signal is thus “compressed” by a non-linear function at the encoder, sent on a transmission channel and “decompressed” by the inverse function at the decoder.
- the advantage of amplitude compression is that it makes it possible to transform the probability distribution of the amplitude of the input audio signal into a quasi-uniform probability law, on which a uniform scalar quantization can be applied.
- the amplitude compression laws are generally laws of the logarithmic type which therefore make it possible to encode a sampled signal with a resolution of 16 bits (in “linear PCM” format) on 8 bits (in “PCM” type A-law format or mu).
- the 8 bits per sample in G.711 are distributed as follows as shown at 15 in FIG.
- an original sample of the signal S to be encoded has an amplitude equal to -75. Therefore, this amplitude is in the range [-80, -65] of line 123 (or "level" 123) of the table.
- the coding of this information consists of delivering a final coded index, referenced I ' MIC in Figure 1 and Table 1, which is equal to 0x51.
- VQ 32256 to all the samples whose initial amplitude was in the interval [31744, 32767], or in all 1024 possible values, which corresponds to a quantization step of 1024.
- SNR signal-to-noise ratio
- the quantization step in the domain of the original signal is proportional to the amplitude of the signals. This signal-to-noise ratio is not enough to make the quantization noise inaudible over the entire frequency band 0-4000 Hz. In addition, for low level signals (which are encoded with the first segment) the SNR is very bad.
- the G.711 standard is generally considered to be of good quality for narrowband telephony applications with terminals limiting the band at [300-3400 Hz]. However, the quality is unsatisfactory when using G.711 for other applications such as for example terminals for good fidelity in the band [50.4000 Hz] or for the expanded bandwidth extension of the G-code. .711.
- Hierarchical coding methods consisting in providing a refinement layer determined from the coding noise of the G.711 coder. This coding noise is then coded by a technique different from G.711, which constitutes the so-called base layer (or core layer).
- G.711 which constitutes the so-called base layer (or core layer).
- Such a hierarchical coding method is for example described in Y. Hiwasaki, H. Ohmuro, T. Mori, S. Kurihara and A. Kataoka. "A G.711 Embedded Wideband Speech Coding for VoIP Conferences", IEICE Trans. Inf. & Syst, Vol. E89-D, No. 9, September 2006.
- This type of method has the disadvantage of significantly increasing the complexity of the encoder while the type of coding MIC is deemed to be of low complexity.
- the PCM coding noise is a white, and therefore uncorrelated noise, the coding of this type of noise is difficult to implement since the compression techniques are essentially based on properties of extraction
- the present invention provides a solution that improves the situation.
- the invention proposes a scalar quantization encoding method of the samples of a digital audio signal, the samples being encoded on a predetermined number of bits to obtain a binary field of quantization indices, the coding being performed according to a law of amplitude compression, a predetermined number of low-order bits not being taken into account in the bit frame of quantization indices.
- the method is such that it comprises the following steps: storing at least a portion of the least significant bits that are not taken into account in the quantization index bit frame;
- an improvement stream is transmitted at the same time as the bit frame of quantization indices.
- This extension stream is determined by taking advantage of low-order bits that are not used during encoding. This method therefore has the advantage of not adding complexity to the encoder and to provide the desired quality improvement by providing the decoder the possibility of obtaining a better decoding accuracy.
- the stored bits are the most significant bits among the bits that are not taken into account in the bit frame of quantization indices.
- the number of bits taken into account to determine the improvement stream is a function of the available bit rate during a transmission to a decoder.
- extension stream is scalable during transmission as a function of the available throughput.
- the invention is particularly suitable in the case where the scalar quantization step is a PCM type quantization according to a logarithmic type A compression compression law or mu type compliant with the ITU-T G.711 standard.
- the invention also applies to a method for decoding a bit frame of quantization indices comprising a predetermined number of bits by an inverse quantization step and according to an amplitude compression law.
- the method is such that it comprises the following steps: receiving an improvement stream comprising one or more extension bits;
- the decoder which receives extension bits, thus improves the accuracy of its expansion or "decompression” by concatenating the received extension bits with those present in the quantization index frame received from the base stream.
- the method further comprises a step of adapting a rounding value according to the number of received extension bits to obtain the decoded audio signal.
- the detection of the coded audio signal is thus adapted as a function of the number of bits of the extension stream.
- the invention also relates to an audio encoder comprising a scalar quantization module of the samples of a digital audio signal, the samples being coded on a predetermined number of bits to obtain a binary field of quantization indices, the coding taking place according to an amplitude compression law, a predetermined number of low-order bits are not taken into account in the bit frame of quantization indices.
- the encoder according to the invention comprises:
- a memory space able to store at least a portion of the low-order bits that are not taken into account in the quantization index bit frame
- the invention relates to an audio decoder adapted to decode a binary field of quantization indices comprising a predetermined number of bits by an inverse quantization module and according to an amplitude compression law.
- the decoder according to the invention comprises:
- the invention finally relates to a computer program intended to be stored in a memory of an encoder and / or a memory medium capable of cooperating with an encoder reader, comprising code instructions for the implementation of the steps of the method of encoding according to the invention when executed by a processor of said encoder.
- the invention aims at a computer program intended to be stored in a memory of a decoder and / or a memory medium capable of cooperating with a decoder reader, comprising code instructions for the implementation of the steps of the decoder. decoding method according to the invention when they are executed by a processor of said decoder.
- FIG. 1 illustrates a conventional G.711 PCM coding / decoding system of the state of the art
- FIG. 2 illustrates a coding / decoding system according to the invention as well as the methods according to the invention, implemented by the elements of this system;
- FIGS. 3 a and 3b show the quantized values with respect to the input values after application of the respective encoding laws A and mu according to the G.711 standard;
- FIGS. 4 and 5 represent a comparison with and without implementation of the invention of the quantized values with respect to the input values after application of the A and mu coding laws, respectively.
- FIG. 2 illustrates a coding / decoding system according to the invention.
- An encoder 23 comprises a quantizer Q MIC 20 capable of quantizing the input signal S to obtain a quantization index frame I M ic which is transmitted on the transmission channel 21 to a decoder 24.
- this encoder is of the PCM encoder type and implements a type A or mu encoding law as described in the G.711 standard.
- the quantization index frame obtained is therefore represented at 15 and conforms to the G.711 law A or mu frame.
- the pseudo code represented in Appendix A-10 gives an example of implementation of Law A as described in the G.711 standard (with a piecewise linear approximation of amplitude compression law).
- a concrete implementation of this pseudo code is also given as an example in Appendix A-10.
- This implementation complies with the ITU-T Recommendation G.191 Sofware Tool Library (STL-2005), Chapter 13 "ITU-T Basic Operators". This recommendation is available on the ITU website: http://www.itu.int/rec/T-REC-G.191-200508-I/en
- the 8-bit quantization index comprises the sign bit, the segment index (exp) and the position on the segment (mant).
- the sign bit is determined which is set to the 0 position as indicated at 15 in FIG. 1.
- the position of the most significant bit "pos" is sought and calculates the segment number which is coded on 3 bits which is put at position 1, 2 and 3 as represented in 15 in FIG.
- the 4 bits constituting the position on the segment are set at positions 4, 5, 6 and 7 as represented at 15.
- MSB most significant bits
- the decoding can be implemented by simple operations as illustrated by the ITU-T STL-2005 pseudo-code and implementation shown in Annex A-I.
- Figures 3a and 3b compare the resolution of these two laws for the first 512 values.
- MSB most significant bits
- the minimum value of the variable "pos" for the coding according to the mu law is 7 since as previously mentioned in the case of the mu law the first segment is treated in the same way as the other segments. There is therefore for all segments at least 3 bits of lower weight that are lost.
- the decoding can be carried out simply by a simple algorithm, an example is given in appendix A-13.
- the encoder 23 takes advantage of the coding method according to the laws A or mu by storing in a memory space represented at 27, a part of the low-order bits which have not been taken into account for the coding of the binary field of quantization indices I MIC -
- the number of bits lost by the A or mu law coding methods increases with the segment number, up to 10 bits for the last segment.
- the method according to the invention makes it possible to recover at least the most significant bits among these lost bits.
- the method according to the invention will store in memory 27, the two most significant bits bits that are not taken into account. in the compression operation to determine the quantization index frame.
- the decoder 24 comprising an inverse quantizer, here an inverse PCM quantizer Q ' Vic 22, receives in parallel the base stream I' MIC and the improvement stream I'EXT-
- I ' MIC and I' E ⁇ are versions possibly disturbed by binary errors of I MIC and I EXT respectively.
- the decoder In case of reception of this improvement stream by the reception means 29 of the decoder 24, the decoder will then have a greater precision on the location of the decoded sample in the segment. For this, it concatenates the extension bits to the bits received in the base stream I ' PCM by means of concatenation bits, and then perform an inverse quantization at 22.
- the improvement stream I EXT consists of two extension bits per sample, ie a bit rate of 16 kbit / s. These extension bits can be obtained by bit shifting in two operations as shown by the pseudo code shown in Appendix A-14.
- the two stored bits are sent in the extension stream.
- the two extension bits are the 8 th and 9 th bit of the compressed signal.
- the pseudo code allowing to do all the operations to the coder for the law A is given in appendix A-15.
- FIG. 4 shows a comparison of the quantized values with respect to the input values between the classical A-law (in dashed line) and the A-law with two bits per sample extension (solid line), for the first 128 values.
- FIG. 5 shows a comparison of the quantized values with respect to the input values between the classical mu law (in dashed line) and the mu law with two bits per sample extension (solid line), for the first 128 values.
- the decoder Upon receipt of the enhancement stream I ' EXT , the decoder concatenates the extension bits thus received behind position bits of the base flow I' MIC to effect the amplitude or expansion decompression which is inverse operation of the amplitude compression process.
- the segment is divided into two. The accuracy on the location in the segment of the decoded value is then greater.
- the Roundval rounding value that finds the value of the middle of the segment is also adapted according to the number of extension bits received.
- the information of the number of extension bits received is for example given via an external signaling as represented by the arrow 26 in FIG. This information could also be deduced directly by the analysis of the extension flow.
- the encoder as represented in FIG. 2 comprises a DSP processor (for "Digital Signal Processor"), not shown here, a memory space 27 for storing at least the bits that will be used to determine the extension stream.
- DSP processor for "Digital Signal Processor”
- memory space 27 for storing at least the bits that will be used to determine the extension stream.
- This memory space 27 may be part of a memory block which furthermore comprises a storage memory and / or a working memory.
- the storage means may comprise a computer program comprising code instructions for implementing the steps of the coding method according to the invention when they are executed by the encoder processor.
- the computer program can also be stored on a memory medium readable by an encoder reader or downloadable in the encoder memory space.
- This encoder thus implements the method according to the invention of coding by scalar quantization of the samples of a digital audio signal.
- the samples are coded on a predetermined number of bits to obtain a bit frame of quantization indices and the coding is performed according to an amplitude compression law. A predetermined number of low-order bits are not taken into account in the bit frame of quantization indices.
- the coding is such that it comprises the following steps:
- the decoder comprises a DSP type processor not shown here and is able to implement the method of decoding a binary quantization index frame comprising a predetermined number of bits by a quantization step inverse and according to a law of amplitude compression. This process is such that it comprises the following steps:
- This decoder further comprises a storage means (not shown) capable of storing a computer program comprising code instructions for implementing the steps of the decoding method according to the invention when they are executed by the decoder processor. .
- the computer program can also be stored on a memory medium readable by a decoder reader or downloadable in the decoder memory space.
- the LSB "ext_bits" of the "ext” variable are sent to the enhancement stream.
- the invention also applies in the case where the transmission rate the flow must be reduced.
- the extension stream comprises two bits, the least significant bit of this extension stream is no longer transmitted.
- the decoder then receives only one extension bit per sample.
- the decoder as described in the exemplary pseudo-code will work correctly with this one bit-per-sample extension layer provided that the received extension bit is set in the variable "ext" to position 1, the position bit 0 of the variable "ext” is then set to 0 and the value of "roundval" is adapted accordingly.
- variable "roundval” as used in the examples given is therefore a function of the number of bits received by the encoder and the law used (A or mu). Table 4 below gives the value of the variable "roundval" in the different situations.
- the increase of the RSB is 12 dB, if a bit is received, the increase of the RSB is 6 dB.
- the encoder can send 4 bits per sample in the extension layer and the decoder can receive 4, 3, 2, 1 or 0 of these bits, the quality of the decoded signal will be proportional with the number of extension bits received.
- the invention will be implemented not according to the algorithms previously specified by pseudo-code but by precalculating and storing in tables at the coder and / or the decoder the levels making it possible to obtain the bits of extension.
- this solution has the disadvantage of requiring greater memory capacity in both the encoder and the decoder for a gain in low complexity.
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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)
Abstract
Description
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Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
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FR0756326 | 2007-07-06 | ||
PCT/FR2008/051248 WO2009010674A1 (en) | 2007-07-06 | 2008-07-04 | Hierarchical coding of digital audio signals |
Publications (2)
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EP2176861A1 true EP2176861A1 (en) | 2010-04-21 |
EP2176861B1 EP2176861B1 (en) | 2013-03-27 |
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EP08806166A Active EP2176861B1 (en) | 2007-07-06 | 2008-07-04 | Hierarchical coding of digital audio signals |
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US (1) | US8577687B2 (en) |
EP (1) | EP2176861B1 (en) |
JP (1) | JP5264901B2 (en) |
KR (1) | KR101476699B1 (en) |
CN (1) | CN101796579B (en) |
ES (1) | ES2416056T3 (en) |
WO (1) | WO2009010674A1 (en) |
Families Citing this family (11)
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KR100912826B1 (en) * | 2007-08-16 | 2009-08-18 | 한국전자통신연구원 | A enhancement layer encoder/decoder for improving a voice quality in G.711 codec and method therefor |
FR2938688A1 (en) * | 2008-11-18 | 2010-05-21 | France Telecom | ENCODING WITH NOISE FORMING IN A HIERARCHICAL ENCODER |
FR2969360A1 (en) * | 2010-12-16 | 2012-06-22 | France Telecom | IMPROVED ENCODING OF AN ENHANCEMENT STAGE IN A HIERARCHICAL ENCODER |
SG188199A1 (en) | 2011-02-22 | 2013-04-30 | Panasonic Corp | Image coding method, image decoding method, image coding apparatus, image decoding apparatus, and image coding and decoding apparatus |
MX2013000093A (en) | 2011-02-22 | 2013-02-27 | Panasonic Corp | Filter method, dynamic image encoding device, dynamic image decoding device, and dynamic image encoding/decoding device. |
FR2981781A1 (en) | 2011-10-19 | 2013-04-26 | France Telecom | IMPROVED HIERARCHICAL CODING |
US10230394B2 (en) * | 2014-09-19 | 2019-03-12 | Telefonaktiebolaget Lm Ericsson (Publ) | Methods for compressing and decompressing IQ data, and associated devices |
EP3051701A1 (en) * | 2015-01-30 | 2016-08-03 | Nxp B.V. | Non-uniform quantization of log-likelihood ratios |
CN104966517B (en) * | 2015-06-02 | 2019-02-01 | 华为技术有限公司 | A kind of audio signal Enhancement Method and device |
CN107680607B (en) * | 2017-09-29 | 2021-05-18 | 联想(北京)有限公司 | Signal compression method, signal decompression method and device thereof |
US11935550B1 (en) * | 2023-03-31 | 2024-03-19 | The Adt Security Corporation | Audio compression for low overhead decompression |
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US3688097A (en) * | 1970-05-20 | 1972-08-29 | Bell Telephone Labor Inc | Digital attenuator for non-linear pulse code modulation signals |
US4386237A (en) * | 1980-12-22 | 1983-05-31 | Intelsat | NIC Processor using variable precision block quantization |
DE3411962A1 (en) * | 1983-03-31 | 1984-10-31 | Sansui Electric Co., Ltd., Tokio/Tokyo | DATA TRANSFER DEVICE |
CA1220282A (en) * | 1985-04-03 | 1987-04-07 | Northern Telecom Limited | Transmission of wideband speech signals |
JP3237178B2 (en) * | 1992-03-18 | 2001-12-10 | ソニー株式会社 | Encoding method and decoding method |
KR100335611B1 (en) * | 1997-11-20 | 2002-10-09 | 삼성전자 주식회사 | Scalable stereo audio encoding/decoding method and apparatus |
KR100335609B1 (en) * | 1997-11-20 | 2002-10-04 | 삼성전자 주식회사 | Scalable audio encoding/decoding method and apparatus |
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US6931058B1 (en) * | 2000-05-19 | 2005-08-16 | Scientific-Atlanta, Inc. | Method and apparatus for the compression and/or transport and/or decompression of a digital signal |
TW501099B (en) * | 1999-08-13 | 2002-09-01 | Koninkl Philips Electronics Nv | Transmission of a digital information signal having M bit PCM samples |
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KR100528327B1 (en) * | 2003-01-02 | 2005-11-15 | 삼성전자주식회사 | Method and apparatus for encoding/decoding audio data with scalability |
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2008
- 2008-07-04 EP EP08806166A patent/EP2176861B1/en active Active
- 2008-07-04 US US12/667,724 patent/US8577687B2/en active Active
- 2008-07-04 WO PCT/FR2008/051248 patent/WO2009010674A1/en active Application Filing
- 2008-07-04 KR KR1020107002702A patent/KR101476699B1/en active IP Right Grant
- 2008-07-04 CN CN200880105867.6A patent/CN101796579B/en active Active
- 2008-07-04 ES ES08806166T patent/ES2416056T3/en active Active
- 2008-07-04 JP JP2010514084A patent/JP5264901B2/en active Active
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WO2009010674A1 (en) | 2009-01-22 |
KR101476699B1 (en) | 2014-12-26 |
US20100191538A1 (en) | 2010-07-29 |
US8577687B2 (en) | 2013-11-05 |
JP2010532876A (en) | 2010-10-14 |
CN101796579B (en) | 2014-12-10 |
JP5264901B2 (en) | 2013-08-14 |
CN101796579A (en) | 2010-08-04 |
EP2176861B1 (en) | 2013-03-27 |
KR20100049579A (en) | 2010-05-12 |
ES2416056T3 (en) | 2013-07-30 |
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