EP2139000A1 - Verfahren und Vorrichtung zur Kodierung und Dekodierung von Sprache bzw. Nicht-Sprache-Audioeingabesignalen - Google Patents

Verfahren und Vorrichtung zur Kodierung und Dekodierung von Sprache bzw. Nicht-Sprache-Audioeingabesignalen Download PDF

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EP2139000A1
EP2139000A1 EP08159018A EP08159018A EP2139000A1 EP 2139000 A1 EP2139000 A1 EP 2139000A1 EP 08159018 A EP08159018 A EP 08159018A EP 08159018 A EP08159018 A EP 08159018A EP 2139000 A1 EP2139000 A1 EP 2139000A1
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speech
signal
encoding
mlt
output
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French (fr)
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EP2139000B1 (de
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Oliver Wuebbolt
Johannes Boehm
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Thomson Licensing SAS
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Deutsche Thomson OHG
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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
    • 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
    • G10L19/0212—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 orthogonal transformation
    • 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
    • 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
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/78—Detection of presence or absence of voice signals

Definitions

  • the invention relates to a method and to an apparatus for encoding or decoding a speech and/or non-speech audio input signal.
  • a disadvantage of the known audio/speech codecs is a clear dependency of the coding quality on the types of content, i.e. music-like audio signals are best coded by audio codecs and speech-like audio signals are best coded by speech codecs.
  • No known codec is holding a dominant position for mixed speech/music content.
  • a problem to be solved by the invention is to provide a good codec performance for both, speech and music, and to further improve the codec performance for such mixed signals. This problem is solved by the methods disclosed in claims 1 and 3. Apparatuses that utilise these methods are disclosed in claims 2 and 4.
  • the inventive joined speech/audio codec uses speech coding techniques as well as audio transform coding techniques.
  • Known transform-based audio coding processing is combined in an advantageous way with linear prediction-based speech coding processing using one or more Modulated Lapped Transform (MLT) at the codec input and one or more inverse Modulated Lapped Transform (IMLT) at the codec output.
  • MLT Modulated Lapped Transform
  • IMLT inverse Modulated Lapped Transform
  • the MLT output spectrum is separated into frequency bins (low frequencies) assigned to the speech coding section of the codec, and the remaining frequency bins (high frequencies) assigned to the transform-based coding section of the codec, wherein the transform length at the codec input and output can be switched signal adaptively.
  • the transform length can be switched input signal adaptively.
  • the invention achieves a uniform good codec quality for both speech-like and music-like audio signals, especially for very low bit rates but also for higher bit rates.
  • the inventive method is suited for encoding a speech and/or non-speech audio input signal, including the steps:
  • the inventive apparatus is suited for encoding a speech and/or non-speech audio input signal, said apparatus including means being adapted for:
  • the inventive method is suited for decoding a bit stream representing an encoded speech and/or non-speech audio input signal that was encoded according to the above method, said decoding method including the steps:
  • the inventive apparatus is suited for decoding a bit stream representing an encoded speech and/or non-speech audio input signal that was encoded according to the above encoding method, said apparatus including means being adapted for:
  • coding processing for speech-like signals linear prediction based speech coding processing, e.g. CELP, ACELP, cf. ISO/IEC 14496-3, Subparts 2 and 3, and MPEG4-CELP
  • state-of-the-art coding processing for general audio or music-like signals based on a time-frequency transform, e.g. MDCT.
  • the PCM audio input signal IS is transformed by a Modulated Lapped Transform MLT having a pre-determined length in step/stage 10.
  • MLT Modulated Lapped Transform
  • a Modified Discrete Cosine Transform MDCT is appropriate for audio coding applications.
  • the MDCT was first called by Princen and Bradley "Oddly-stacked Time Domain Alias Cancellation Transform" and was published in John P. Princen and Alan B. Bradley, "Analysis/synthesis filter bank design based on time domain aliasing cancellation", IEEE Transactions on Acoustics Speech Sigal Processing ASSP-34 (5), pp.1153-1161, 1986 .
  • the obtained spectrum is separated into frequency bins belonging to the speech band (representing a low band signal) and the remaining bins (high frequencies) representing a remaining band signal RBS.
  • the speech band bins are transformed back into time domain using the inverse MLT, e.g. an inverse MDCT, with a short transform length with respect to the pre-determined length used in step/stage 10.
  • the resulting time signal has a lower sampling frequency than the input time signal and contains only the corresponding frequencies of the speech band bins.
  • the generated time domain signal is then used as input signal for a speech encoding step/stage 12.
  • the output of the speech encoding can be transmitted in the output bit stream OBS, depending on a decision made by a below-described speech/audio switch 15.
  • the encoded 'speech' signal is decoded in a related speech decoding step/stage 13, and the decoded 'speech' signal is transformed back into frequency domain in step/stage 14 using the MLT corresponding to the inverse MLT of step/stage 11 (i.e. an 'opposite type' MLT having the short length) in order to re-generate the speech band signal, i.e. a reconstructed speech signal RSS.
  • That switch it is decided, whether the original low frequency bins are coded together with the remaining high frequency bins (this indicates that the coded 'speech' signal is not transmitted in bit stream OBS), or the difference signal DS is coded together with the remaining high frequency bins in a following quantisation&coding step/stage 16 (this indicates that the coded 'speech' signal is transmitted in bit stream OBS).
  • That switch may be operated by using a rate-distortion optimisation.
  • An information item SWI about the decision of switch 15 is included in bit stream OBS for use in the decoding.
  • step/stage 16 In this switch, but also in the other steps/stages, the different delays introduced by the cascaded transforms are to be taken into account.
  • the different delays can be balanced using corresponding buffering for these steps/stages. It is possible to use a mixture of original frequency bins and difference signal frequency bins in the low frequency band as input to step/stage 16. In such case, information about how that mixture is composed is conveyed to the decoding side.
  • the remaining frequency bins output by step/stage 10 i.e. the high frequencies
  • step/stage 16 an appropriate quantisation is used (e.g. like the quantisation techniques used in AAC), and subsequently the quantised frequency bins are coded using e.g. Huffman coding or arithmetic coding.
  • the speech/audio switch 15 decides that a music-like input signal is present and therefore the speech coder/decoder or its output is not used at all, the original frequency bins corresponding to the speech band are to be encoded (together with the remaining frequency bins) in the quantisation&coding step/stage 16.
  • the quantisation&coding step/stage 16 is controlled by a psycho-acoustic model calculation 18 that exploits masking properties of the input signal IS for the quantisation. Therefore side information SI can be transmitted in the bit stream multiplex to the decoder.
  • Switch 15 can also receive suitable control information (e.g. degree of tonality or spectral flatness, or how noise-like the signal is) from psycho-acoustic model step/stage 18.
  • a bit stream multiplexer step/stage 17 combines the output code (if present) of the speech encoder 12, the switch information of switch 15, the output code of the quantisation&coding step/stage 16, and optionally side information code SI, and provides the output bit stream OBS.
  • inverse MDCT inverse MDCT
  • iMDCT inverse MDCT
  • the inverse MLT steps/stages 22 are arranged between a first grouping step/stage 21 and a second grouping step/stage 23 and provide a doubled number of output values.
  • the number of combined MLT bins which means the transform length of the inverse MLT, defines the resulting time and frequency resolution, wherein a longer inverse MLTs delivers a higher time resolution.
  • overlap/add is performed (optionally involving application of window functions) and the output of the inverse MLTs applied on the same input spectrum is sorted such that it results in several (the quantity depends on the size of the inverse MLTs) temporally successive 'short block' spectra which are quantised and coded in step/stage 16.
  • the information about this 'short block coding' mode being used is included in the side information SI.
  • multiple 'short block coding' modes with different inverse MLT transform lengths can be used and signalled in SI.
  • a non-uniform time-frequency resolution over the short block spectra is facilitated, e.g. a higher time resolution for high frequencies and a higher frequency resolution for low frequencies.
  • the inverse MLT can get a length of 2 successive frequency bins and for the highest frequencies the inverse MLT can get a length of 16 successive frequency bins.
  • a non-uniform frequency resolution is chosen, it is not possible to group e.g. 8 short block spectra.
  • a different order of coding the resulting frequency bins can be used, for example one 'spectrum' may contain not only different frequency bins at a time, but also the same frequency bins at different points in time may be included.
  • the input signal IS adaptive switching between the processing according to Fig. 1 and the processing according to Fig. 2 is controlled by psycho-acoustic model step/stage 18. For example, if from one frame to the following frame the signal energy in input signal IS rises above a threshold (i.e. there is a transient in the input signal), the processing according to Fig. 2 is carried out. In case the signal energy is below that threshold, the processing according to Fig. 1 is carried out.
  • This switching information is included in output bitstream OBS for a corresponding switching in the decoding.
  • the transform block sections can be weighted by a window function, in particular in an overlapping manner, wherein the length of a window function corresponds to the current transform length.
  • Analysis and synthesis windows can be identical, but need not.
  • a further window function is disclosed in table 7.33. of the AC-3 audio coding standard.
  • transition window functions are used, e.g. as described in B.Edler, "Cod mich von Audiosignalen mit überlappender Transformation und adaptiven Novafunktionen", FREQUENZ, vol.43, pp.252-256, 1989 , or as used in mp3 and described in the MPEG1 standard ISO/IEC 11172-3 in particular section 2.4.3.4.10.3, or as in AAC (e.g. as described in the MPEG4 standard ISO/IEC 14496-3, Subpart 4).
  • the received or replayed bit stream OBS is demultiplexed in a corresponding step/stage 37, thereby providing code (if present) for the speech decoder 33, the switch information SWI for switch 35, the code and the switching information for the decoding step/stage 36, and optionally side information code SI.
  • code if present
  • the speech subcoder 11,12,13,14 was used at encoding side for a current data frame, in that current frame the corresponding encoded speech band frequency bins are correspondingly reconstructed by the speech decoding step/stage 33 and the downstream MLT step/stage 34, thereby providing the reconstructed speech signal RSS.
  • the remaining encoded frequency bins are correspondingly decoded in decoding step/stage 36, whereby the encoder-side quantisation operation is reversed correspondingly.
  • the speech/audio switch 35 operates corresponding to its operation at encoding side, controlled by switch information SWI.
  • switch information SWI indicates that a music-like input signal is present in the current frame and therefore the speech coding/decoding was not used
  • the frequency bins corresponding to the low band are decoded together with the remaining frequency bins in the decoding step/stage 36, thereby providing the reconstructed remaining band signal RRBS and the reconstructed low band signal RLBS.
  • the output signal or signals of step/stage 36 and of switch 35 are correspondingly combined in inverse MLT (e.g.
  • iMDCT iMDCT
  • switch 35 and in the other steps/stages the different delays introduced by the cascaded transforms are to be taken into account.
  • the different delays can be balanced using corresponding buffering for these steps/stages.
  • the corresponding option was used at encoding side, not the frequency bins of the combined signal CS, but the frequency bins of the reconstructed speech signal RSS are used for the corresponding processing in switch 35 and in step/stage 30, i.e. in step/stages 16 and 36, respectively, there is no coding/decoding at all of the low band spectrum.
  • step/stage 36 of the 'short block mode' is illustrated in Fig. 4 .
  • the decoding in step/stage 36 of the 'short block mode' is illustrated in Fig. 4 .
  • several temporally successive 'short block' spectra are to be decoded in step/stage 36 and collected in a first grouping step/stage 43. Overlap/add is performed (optionally involving application of window functions). Thereafter each set of temporally successive spectral coefficients is transformed using the corresponding MLT steps/stages 42, and provides a halved number of output values.
  • the generated spectral coefficients are then grouped in a second grouping step/stage 41 to one MLT spectrum with the initial high frequency resolution and transform length.
  • multiple 'short block decoding' modes with different MLT transform lengths can be used as signalled in SI, whereby a non-uniform time-frequency resolution over the short block spectra is facilitated, e.g. a higher time resolution for high frequencies and a higher frequency resolution for low frequencies.
  • a different cascading of the MLTs can be used wherein the order of the inner MLT/inverse MLT pair in the speech encoder is switched.
  • Fig. 5 a block diagram of a corresponding encoding is depicted, wherein Fig. 1 reference signs mean the same operation as in Fig. 1 .
  • the inverse MLT 11 is replaced by an MLT step/stage 51
  • the MLT 14 is replaced by an inverse MLT step/stage 54 (i.e. an 'opposite type' MLT). Due to the exchanged order of these MLTs the speech encoder input signal has different properties compared to those in Fig. 1 . Therefore the speech coder 52 and the speech decoder 53 are adapted to these different properties (e.g. such that aliasing components are cancelled out).
  • a 'short block mode' processing can be used as shown in Fig. 6 , wherein MLT steps/stages 62 corresponding to that in Fig. 4 replace the inverse MLT steps/stages 22 in Fig. 2 .
  • the speech decoding step/stage 33 in Fig. 3 is replaced by a correspondingly adapted speech decoding step/stage 73 and the MLT step/stage 34 in Fig. 3 is replaced by a corresponding inverse MLT step/stage 74.
  • a 'short block mode' processing can be used as shown in Fig. 8 , wherein corresponding inverse MLT steps/stages 82 corresponding to that in Fig. 1 replace the MLT steps/stages 42 in Fig. 4 .
  • a different way of block switching is carried out.
  • a fixed large MLT 10 e.g. an MDCT
  • several short MLTs (or MDCTs) 90 can be switched on.
  • a fixed large MLT 10 e.g. an MDCT
  • 8 short MDCTs with a transform length of 256 samples can be used.
  • the sum of the lengths of the short transforms is equal to the long transform length (although it makes buffer handling even more easier).
  • the internal buffer handling is easier than for the long/short block mode switching according to figures 1 to 8 , at the cost of a less sharp band separation between the speech frequency band and the remaining frequency band.
  • the reason for the internal buffer handling being easier is as follows: at least for each inverse MLT operation an additional buffer is required, which leads in case of an inner transform to the necessity of an additional buffer also in the parallel high frequency path. Therefore the switching at the outmost transform has the least side effects concerning buffers.
  • the short blocks are used only for encoding transient input signals, the sharp separation in time domain is more important.
  • Fig. 9 the Fig. 1 reference signs do mean the same operation as in Fig. 1 .
  • the MLT 10 is input signal IS adaptively replaced by short MLT steps/stages 90, the inverse MLT 11 is replaced by shorter inverse MLT steps/stages 91, and the MLT 14 is replaced by shorter MLT steps/stages 94. Due to this kind of blocks switching, the lengths of the first transform 90, 30 and the second transform 11, 34, 51, 74 (iMDCT to reconstruct the speech band) and the third transform 14, 54 are coordinated. Furthermore, several short blocks of the speech band signal can be buffered after the iMDCT 91 in Fig. 9 in order to collect enough samples for a complete input frame for the speech coder.
  • the encoding of Fig. 9 can also be adapted correspondingly to the encoding described for Fig. 5 .
  • the decoding according to Fig. 3 is adapted correspondingly, i.e. the inverse MLTs 34 and 30 are each replaced by corresponding adaptively switched shorter inverse MLTs.
  • the transform block sections are weighted at encoding side in MLT 90 and at decoding side in inverse MLT 30 by window functions, in particular in an overlapping manner, wherein the length of a window function corresponds to the current transform length. In case of switching the transform length, to achieve a smooth transition between long and short blocks, especially shaped long windows (the start and stop windows, or transition windows) are used.

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EP08159018A 2008-06-25 2008-06-25 Verfahren und Vorrichtung zur Kodierung und Dekodierung von Sprache bzw. Nicht-Sprache-Audioeingabesignalen Not-in-force EP2139000B1 (de)

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CN2009101503026A CN101615393B (zh) 2008-06-25 2009-06-19 对语音和/或非语音音频输入信号编码或解码的方法和设备

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CN102737636A (zh) * 2011-04-13 2012-10-17 华为技术有限公司 一种音频编码方法及装置
CN102737636B (zh) * 2011-04-13 2014-06-04 华为技术有限公司 一种音频编码方法及装置
RU2667380C2 (ru) * 2014-06-24 2018-09-19 Хуавэй Текнолоджиз Ко., Лтд. Способ и устройство кодирования аудио
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US11074922B2 (en) 2014-06-24 2021-07-27 Huawei Technologies Co., Ltd. Hybrid encoding method and apparatus for encoding speech or non-speech frames using different coding algorithms
CN106033982A (zh) * 2015-03-13 2016-10-19 中国移动通信集团公司 一种实现超宽带语音互通的方法、装置和终端
CN106033982B (zh) * 2015-03-13 2018-10-12 中国移动通信集团公司 一种实现超宽带语音互通的方法、装置和终端

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