WO2003042976A1 - Procede et systeme de traitement de signaux audio - Google Patents

Procede et systeme de traitement de signaux audio Download PDF

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Publication number
WO2003042976A1
WO2003042976A1 PCT/IB2002/004539 IB0204539W WO03042976A1 WO 2003042976 A1 WO2003042976 A1 WO 2003042976A1 IB 0204539 W IB0204539 W IB 0204539W WO 03042976 A1 WO03042976 A1 WO 03042976A1
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WO
WIPO (PCT)
Prior art keywords
processing
audio signals
audio
quality
received audio
Prior art date
Application number
PCT/IB2002/004539
Other languages
English (en)
Inventor
Erik Larsen
Jo Smeets
Stefan M. J. Willems
Original Assignee
Koninklijke Philips Electronics N.V.
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Koninklijke Philips Electronics N.V. filed Critical Koninklijke Philips Electronics N.V.
Publication of WO2003042976A1 publication Critical patent/WO2003042976A1/fr

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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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
    • G10L21/0364—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude for improving intelligibility
    • 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

Definitions

  • the present invention relates to a method for processing audio signals and to an audio processing system for applying this method.
  • Audio signals may be transmitted electronically, for example, over internet.
  • the audio signals may be transmitted in a compressed form, for example, MP3, MP3Pro, WMA or Real Audio format, for reasons of reduced need of transmission bandwidth.
  • the compression factor may be variable, leading to a variety of audio signal stream bitrates, for example from 16 kbit/s up to 196 kbit/s and sample frequencies from 8 kHz up to 48 kHz.
  • the decoded audio signals are not perceptually identical to the original source material. Typically, for low bitrates, high frequencies are missing. Possibly, the audio signals are converted to monosignals, whereas the originals are stereosignals.
  • bandwidth extensions methods like described in PHNLO 10607, can re-introduce some missing high frequencies.
  • the problem is that because of the varying quality of the received audio signal stream, due to varying sample frequencies and bitrates, the processing methods must be adaptable, to be able to adjust to the received audio signal quality and to deliver a consistent higher quality after processing.
  • a method for processing audio signals in which the extent of processing is adapted depending on the quality of the received audio signals.
  • the quality determination can be realized in two different manners: the received audio signals are decoded and processed, while the processing is adapted depending on the quality of the received audio signals, as determined before decoding or adapted depending on the quality of the received audio signals, as determined after decoding.
  • the proposed methods enable that the output audio signal stream has a higher audio signal quality than the input audio signal quality, if the processing improving the quality depends on characteristics of the audio input signals.
  • An other choice could be to adapt the processing such that the audio output signals has a constant quality.
  • the adaptability used here is different from the scalability proposed in PHNL000610.
  • QoS quality of service
  • the scalability referred to is to be seen in the context of a resource-limited realtime application, possibly running several algorithms at once.
  • the algorithms are scalable in the sense that resources required by them individually are variable at the expense of lower performance quality, so that the total amount of resources required is controlled and restricted, independent of the number of algorithms requiring these resources.
  • the adaptability according to the invention refers to an adaptation in the parameters of the algorithm, not in output quality or required resources.
  • quality is referred to any aspect of the audio signal input that has a perceptual influence. It is this distinction which supports the present invention.
  • Adaptation of the audio signal processing can be performed in three different senses. Firstly, depending on the quality of the audio signals the processing may be switched on or off; this means that audio signals are fully processed or not at all. Secondly, depending on the quality of the audio signals a parameter may be derived to indicate the amount of processing; this means that audio signals are processed to some extent between not and fully processed. Finally, depending on the quality of the audio signals the processing may adapt to one or more characteristics of the audio input signal; this means that the audio signals are continuously processed but that the processing is directed to a specific parameter of the audio signals.
  • the adaptation of the audio signal processing can be applied in the conversion of the number of audio channels; from N channels in the audio signal stream to M channels available for playback.
  • N may be used, as described in, for example, US-A-6,292,570, relating to a surround sound audio system, PHN 17.814, relating to a headphone system, and PHNL010164, relating to a 3D sound system with headphones.
  • the quality of the audio stream is in this case simply the number of channels which is stripped from the audio packets before decoding. Since N is fixed per repertoire or broadcast, and M is fixed until the user changes his reproduction set-up, this system is adaptable in the first sense. If N ⁇ M a multi-channel decoder will be used, as described in, for example, PHN000708, relating to a multi-channel audio converter; further the same comments apply as in the case N>M.
  • the adaptation of the audio signal processing can be applied in spaciousness: decorrelators, as described, for example, in US-A-6,084,970, relating to a mono-stereo conversion device, an audio reproduction system using such a conversion device and a mono-stereo conversion method, can be used to convert mono to stereo sound (a parameterizable version thereof). -This is adaptable in the first or second sense.
  • the "extent" of mono components in the audio signals is determined by comparing for examples envelopes in left and right channels, RMS errors between left and right channels, and so on.
  • Stereo base widening systems as described in, for example, US-A-5,742,687, relating to stereophonic audio reproduction can be used to increase the stereobase (apparent loudspeaker separation). This is adaptable in the first or second sense too. The extent to which this processing is applied, is determined in the same fashion as for enormous mono sound.
  • the adaptation of the audio signal processing can further be applied in bandwidth extension. Therefore systems as described in, for example, PHNL000249, relating to an infra bass system with variable sub-harmonics, or PHNLO 10607, mentioned before, can be used. These systems are adaptable in the third sense, since depending on the input bandwidth, filter characteristics are adapted. Filter coefficients for different frequency range can be stored in a memory or calculated on-the-fly.
  • the input audio bandwidth is determined, for example, either from bit-rate and/or sample frequency information stripped from the headers of the audio packets or by analyzing frequency content of the decoded audio signals. This analysis can be done in a filter bank where signal energy per frequency range is determined. In another embodiment, bandwidth determination may be done on subband coded signals by analyzing energy distribution per subband. This leads directly to information about the audio spectrum of the signals.
  • the invention does not only relate to a method for processing audio signals, but also to an audio processing system, comprising decoding and processing means for decoding and processing received audio signals.
  • the audio processing system is characterized in that a quality determination unit is provided by means of which, depending on the quality of the supplied audio signals, the extent of processing in the processing means is controlled.
  • the invention further relates to an algorithm for processing the audio signals in said audio processing system; this algorithm may also be applied in the above method for processing audio signals.
  • the invention also relates to an audio apparatus, provided with the above audio processing system, to a computer program capable of running on signal processing means in such an audio apparatus or cooperating with such an audio apparatus comprising the above audio processing system and to an information carrier, carrying instructions to be executed by signal processing means, the instructions being such as to enable said signal processing means to perform the above method for processing audio signal.
  • the present invention may be applied in all sound reproduction systems, specifically in mp3 players and the like, internet radio and fixed and mobile communication systems.
  • FIGs. 1 and 2 show basic block diagrams for an audio processing system according to the invention
  • Fig. 3 shows the application of the method for processing audio signals in a system for adapting stereo channels
  • Fig. 4 shows the application of the method for processing audio signals in a system for the extension of the bandwidth of audio signals.
  • the basic block diagrams of figs. 1 and 2 show a decoder 1, to which received audio input signals are supplied and a processing unit 2 for processing the decoded audio signals. Further there is provided a quality determining unit 3A (fig. 1) and 3B (fig. 2) to adapt the processing in the processing unit 2.
  • the input signals of the quality determining unit can be formed by the received audio input signals before they are decoded (fig. 1) or by the received audio signals after they are decoded (fig. 2).
  • Fig. 3 shows the application of the audio processing system of fig. 2 for the adaptation of stereo signals.
  • Left and right audio signals, L and R respectively, are each supplied via a controllable amplification device 4 and via a controllable amplification device 5 and a filter 6 A and 6B respectively to a combination device 7.
  • the filters 6 A and 6B form complementary comb filters.
  • the control devices 4 and 5 are adjusted by the quality determining unit 3B.
  • L ⁇ R, 0 ⁇ o ⁇ l. If a 0, the original spatial sound effect of the different audio signals L and R is maintained.
  • an audio signal splitting unit 8 is indicated in dashed lines; the single-channel signal M is splitted into signals L and R which are subjected to complementary filtering.
  • L and R are real stereo signals, no filtering needs to take place; in this case the audio processing system is adaptable in the first sense. Nevertheless, it is possible in case L and R are real stereo signals and L ⁇ R, to choose ⁇ O. By means of a suitable determination of the similarity between L and R, 0 ⁇ l. In this case, the audio signal will attain a suitable spatial effect.
  • an audio processing system adaptable in the second sense is obtained, when a can take values between 0 and 1.
  • Fig. 4 shows the application of the audio processing system of fig. 2 for the extension of the bandwidth of audio signals.
  • Audio signals with a bandwidth of, for example, 0 to f ⁇ are supplied on the one hand via the following processing means: a bandfilter 9, a nonlinear element 10, a bandfilter 11 and an amplification device 12 to a combination device 13 and on the other hand via a delay device 14 to said combination device 13.
  • the bandfilters 9 and 11, the amplification device 12 and the delay device 14 are adjusted by the quality determining unit 3B, in this case a bandwidth determination unit, such that the frequencies passed are in an interval of V 2 f ⁇ to fi .
  • the bandfilter 11 is also controlled by the quality determining unit 3B.
  • the supplied audio signals are combined after a delay, corresponding with the total delay of all filters in the processing means, with the processed audio signals such that audio signals with a frequency band from 0 to 2 are obtained.
  • the frequency band is extended at the high frequency side.
  • a system for extending the bandwidth a the low frequency side is also possible.
  • an upsampler precedes the processing unit (before the signal is split as indicated in fig. 4).
  • Such an upsampler will only be used if needed and is thus also controlled by the quality determination unit 3 A, 3B.
  • the quality determining unit 3B is provided with the decoded audio signals, this quality determining unit may also be provided with coded audio packets in correspondence with the basic system of fig 1.
  • An application based on the system of fig. 1 is the case of transmitting audio signals via internet.
  • audio packets such as audio internet packets (JP)
  • JP audio internet packets
  • the audio packets are processed after being decoded.
  • the bandfilters 9 and 11 are adjusted in accordance with the invention.
  • the embodiments described above may be realized by an algorithm, at least part of which may be in the form of a computer program capable of running on signal processing means in an audio reproducing apparatus.
  • a computer program capable of running on signal processing means in an audio reproducing apparatus.
  • these units can be considered as subparts of the computer program.

Landscapes

  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (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)
  • Stereophonic System (AREA)

Abstract

L'invention concerne un procédé de traitement de signaux audio. Les signaux audio reçus sont décodés et traités, la portée du traitement étant adaptée en fonction de la qualité de ces signaux déterminée soit avant, soit après le décodage. On effectue le réglage du traitement en activant ou en désactivant ledit traitement, en dérivant un paramètre qui détermine la portée du traitement entre aucun traitement ou traitement complet ou en continuant à régler le traitement sur la base de certaines caractéristiques des signaux audio reçus.
PCT/IB2002/004539 2001-11-16 2002-10-28 Procede et systeme de traitement de signaux audio WO2003042976A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
EP01204390 2001-11-16
EP01204390.7 2001-11-16

Publications (1)

Publication Number Publication Date
WO2003042976A1 true WO2003042976A1 (fr) 2003-05-22

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PCT/IB2002/004539 WO2003042976A1 (fr) 2001-11-16 2002-10-28 Procede et systeme de traitement de signaux audio

Country Status (1)

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WO (1) WO2003042976A1 (fr)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2347585A1 (fr) * 2008-10-13 2011-07-27 General instrument Corporation Sélection d'un mode adaptateur et communication de données sur la base du mode adaptateur sélectionné

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0664536A1 (fr) * 1994-01-24 1995-07-26 Nokia Mobile Phones Ltd. Procédé pour coder le langage
US6011846A (en) * 1996-12-19 2000-01-04 Nortel Networks Corporation Methods and apparatus for echo suppression
EP1154408A2 (fr) * 2000-05-10 2001-11-14 Kabushiki Kaisha Toshiba Codage de parole et réduction de bruit multimode

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0664536A1 (fr) * 1994-01-24 1995-07-26 Nokia Mobile Phones Ltd. Procédé pour coder le langage
US6011846A (en) * 1996-12-19 2000-01-04 Nortel Networks Corporation Methods and apparatus for echo suppression
EP1154408A2 (fr) * 2000-05-10 2001-11-14 Kabushiki Kaisha Toshiba Codage de parole et réduction de bruit multimode

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2347585A1 (fr) * 2008-10-13 2011-07-27 General instrument Corporation Sélection d'un mode adaptateur et communication de données sur la base du mode adaptateur sélectionné
EP2347585A4 (fr) * 2008-10-13 2013-12-25 Motorola Mobility Llc Sélection d'un mode adaptateur et communication de données sur la base du mode adaptateur sélectionné

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