WO2006091150A1 - Lissage par filtre ameliore dans le codage et/ou le decodage audio multicanaux - Google Patents

Lissage par filtre ameliore dans le codage et/ou le decodage audio multicanaux Download PDF

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Publication number
WO2006091150A1
WO2006091150A1 PCT/SE2006/000234 SE2006000234W WO2006091150A1 WO 2006091150 A1 WO2006091150 A1 WO 2006091150A1 SE 2006000234 W SE2006000234 W SE 2006000234W WO 2006091150 A1 WO2006091150 A1 WO 2006091150A1
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filter
signal
encoding
smoothing
performance
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PCT/SE2006/000234
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WO2006091150B1 (fr
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Anisse Taleb
Stefan Andersson
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Telefonaktiebolaget Lm Ericsson (Publ)
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Priority claimed from PCT/SE2005/002033 external-priority patent/WO2006091139A1/fr
Application filed by Telefonaktiebolaget Lm Ericsson (Publ) filed Critical Telefonaktiebolaget Lm Ericsson (Publ)
Priority to ES06716924T priority Critical patent/ES2389499T3/es
Priority to CN2006800056513A priority patent/CN101128867B/zh
Priority to EP06716924A priority patent/EP1851759B1/fr
Publication of WO2006091150A1 publication Critical patent/WO2006091150A1/fr
Publication of WO2006091150B1 publication Critical patent/WO2006091150B1/fr

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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
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • 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 generally relates to audio encoding and decoding techniques, and more particularly to multi-channel audio encoding/decoding such as stereo coding/decoding.
  • FIG. 1 A general example of an audio transmission system using multi-channel coding and decoding is schematically illustrated in Fig. 1.
  • the overall system basically comprises a multi-channel audio encoder 100 and a transmission module 10 on the transmitting side, and a receiving module 20 and a multi-channel audio decoder 200 on the receiving side.
  • the simplest way of stereophonic or multi-channel coding of audio signals is to encode the signals of the different channels separately as individual and independent signals, as illustrated in Fig. 2. However, this means that the redundancy among the plurality of channels is not removed, and that the bit-rate requirement will be proportional to the number of channels.
  • Another basic way used in stereo FM radio transmission and which ensures compatibility with legacy mono radio receivers is to transmit a sum and a difference signal of the two involved channels.
  • M/S stereo coding is similar to the described procedure in stereo FM radio, in a sense that it encodes and transmits the sum and difference signals of the channel sub-bands and thereby exploits redundancy between the channel sub-bands.
  • the structure and operation of a coder based on M/S stereo coding is described, e.g. in reference [I].
  • Intensity stereo on the other hand is able to make use of stereo irrelevancy. It transmits the joint intensity of the channels (of the different sub-bands) along with some location information indicating how the intensity is distributed among the channels. Intensity stereo does only provide spectral magnitude information of the channels, while phase information is not conveyed. For this reason and since temporal inter-channel information (more specifically the inter-channel time difference) is of major psycho- acoustical relevancy particularly at lower frequencies, intensity stereo can only be used at high frequencies above e.g. 2 kHz. An intensity stereo coding method is described, e.g. in reference [2].
  • Binaural Cue Coding (BCC) is described in reference [3].
  • BCC Binaural Cue Coding
  • This method is a parametric multi-channel audio coding method.
  • the basic principle of this kind of parametric coding technique is that at the encoding side the input signals from N channels are combined to one mono signal.
  • the mono signal is audio encoded using any conventional monophonic audio codec.
  • parameters are derived from the channel signals, which describe the multichannel image.
  • the parameters are encoded and transmitted to the decoder, along with the audio bit stream.
  • the decoder first decodes the mono signal and then regenerates the channel signals based on the parametric description of the multi-channel image.
  • BCC Binaural Cue Coding
  • the principle of the Binaural Cue Coding (BCC) method is that it transmits the encoded mono signal and so-called BCC parameters.
  • the BCC parameters comprise coded inter-channel level differences and inter-channel time differences for sub-bands of the original multi-channel input signal.
  • the decoder regenerates the different channel signals by applying sub-band-wise level and phase and/or delay adjustments of the mono signal based on the BCC parameters.
  • M/S or intensity stereo is that stereo information comprising temporal inter-channel information is transmitted at much lower bit rates.
  • BCC is computationally demanding and generally not perceptually optimized.
  • the side information consists of predictor filters and optionally a residual signal.
  • the predictor filters estimated by an LMS algorithm, when applied to the mono signal allow the prediction of the multi-channel audio signals. With this technique one is able to reach very low bit rate encoding of multi-channel audio sources, however at the expense of a quality drop.
  • Fig. 3 displays a layout of a stereo codec, comprising a down-mixing module 120, a core mono codec 130, 230 and a parametric stereo side information encoder/decoder 140, 240.
  • the down-mixing transforms the multi-channel (in this case stereo) signal into a mono signal.
  • the objective of the parametric stereo codec is to reproduce a stereo signal at the decoder given the reconstructed mono signal and additional stereo parameters.
  • This technique synthesizes the right and left channel signals by filtering sound source signals with so-called head-related filters.
  • this technique requires the different sound source signals to be separated and can thus not generally be applied for stereo or multi-channel coding.
  • the present invention overcomes these and other drawbacks of the prior art arrangements.
  • Another particular object of the invention is to provide a method and apparatus for decoding an encoded multi-channel audio signal.
  • Yet another particular object of the invention is to provide an improved audio transmission system.
  • the invention relies on the basic principle of encoding a first signal representation of one or more of the multiple channels in a first encoding process, and encoding a second signal representation of one or more of the multiple channels in a second, filter-based encoding process.
  • a general inventive concept of the invention is therefore to perform signal-adaptive filter smoothing in the second, filter-based encoding process or in the corresponding decoding process.
  • the signal-adaptive filter smoothing is based on the procedure of estimating expected performance of the first encoding process and/or the second encoding process, and dynamically adapting the filter smoothing in dependence on the estimated performance.
  • the filter smoothing it is possible to more flexibly control the filter smoothing so that it is performed only when really needed. Consequently, unnecessary reduction of the signal energy, for example when the expected coding performance is sufficient, can be avoided completely.
  • the filter smoothing dependent on characteristics of the multi-channel audio input signal, such as inter-channel correlation characteristics, it is possible to first estimate the expected performance of the encoding process(es) and then adjust the degree and/or type of smoothing accordingly.
  • the first encoding process may be a main encoding process and the first signal representation may be a main signal representation.
  • the second encoding process may for example be an auxiliary/side signal process, and the second signal representation may then be a side signal representation such as a stereo side signal.
  • the performance of a filter of the second encoding process is estimated based on characteristics of the multi-channel audio signal, and the filter smoothing is then preferably adapted in dependence on the estimated filter performance of the second encoding process.
  • the filter smoothing is performed by modifying the filter in dependence on the estimated filter performance. This normally involves reducing the energy of the filter.
  • an adaptive smoothing factor is determined in dependence on the estimated filter performance, and the filter is modified by means of the adaptive smoothing factor.
  • the filter smoothing may be based on estimated expected performance of the second encoding process in general, and based on the ICP filter performance in particular.
  • the ICP filter performance is typically representative of the prediction gain of the inter-channel prediction.
  • the signal-adaptive filter smoothing proposed by the invention can be performed on the decoding side.
  • the decoding side is responsive to information representative of signal-adaptive filter smoothing from the encoding side, and performs signal-adaptive filter smoothing in a corresponding second decoding process based on this information.
  • the signal-adaptive information comprises a smoothing factor that depends on estimated performance of an encoding process on the encoding side.
  • Fig. 1 is a schematic block diagram illustrating a general example of an audio transmission system using multi-channel coding and decoding.
  • Fig. 2 is a schematic diagram illustrating how signals of different channels are encoded separately as individual and independent signals.
  • Fig. 3 is a schematic block diagram illustrating the basic principles of parametric stereo coding.
  • Fig. 4 is a diagram illustrating the cross spectrum of mono and side signals.
  • Fig. 5 is a schematic block diagram of a multi-channel encoder according to an exemplary preferred embodiment of the invention.
  • Fig. 6 is a schematic flow diagram setting forth a basic multi-channel encoding procedure according to a preferred embodiment of the invention.
  • Fig. 7 is a more detailed schematic flow diagram illustrating an exemplary encoding procedure according to a preferred embodiment of the invention.
  • Fig. 8 is a schematic block diagram illustrating relevant parts of an encoder according to an exemplary preferred embodiment of the invention.
  • Fig. 9 is a schematic block diagram illustrating relevant parts of a side encoder and an associated control system according to an exemplary embodiment of the invention.
  • Fig. 10 illustrates relevant parts of a decoder according to preferred exemplary embodiment of the invention.
  • the invention relates to multi-channel encoding/decoding techniques in audio applications, and particularly to stereo encoding/decoding in audio transmission systems and/or for audio storage.
  • Examples of possible audio applications include phone conference systems, stereophonic audio transmission in mobile communication systems, various systems for supplying audio services, and multi-channel home cinema systems.
  • BCC on the other hand is able to reproduce the stereo or multi-channel image even at low frequencies at low bit rates of e.g. 3 kbps since it also transmits temporal inter- channel information.
  • this technique requires computationally demanding time-frequency transforms on each of the channels both at the encoder and the decoder.
  • BCC does not attempt to find a mapping from the transmitted mono signal to the channel signals in a sense that their perceptual differences to the original channel signals are minimized.
  • the LMS technique also referred to as inter-channel prediction (ICP), for multi- channel encoding, see [4], allows lower bit rates by omitting the transmission of the residual signal.
  • ICP inter-channel prediction
  • an unconstrained error minimization procedure calculates the filter such that its output signal matches best the target signal.
  • several error measures may be used.
  • the mean square error or the weighted mean square error are well known and are computationally cheap to implement.
  • the accuracy of the ICP reconstructed signal is governed by the present inter-channel correlations.
  • Bauer et al. [8] did not find any linear relationship between left and right channels in audio signals.
  • strong inter-channel correlation is found in the lower frequency regions (0 — 2000 Hz) for speech signals.
  • the ICP filter as means for stereo coding, will produce a poor estimate of the target signal.
  • BCC uses overlapping windows in both analysis and synthesis.
  • coding artifacts introduced by ICP filtering are perceived as more annoying than temporary reduction in stereo width. It has been recognized that the artifacts are especially annoying when the coding filter provides a poor estimate of the target signal; the poorer the estimate, the more disturbing artifacts. Therefore, a basic idea according to the invention is to introduce signal-adaptive filter smoothing as a new general concept for solving the problems of the prior art.
  • Fig. 5 is a schematic block diagram of a multi-channel encoder according to an exemplary preferred embodiment of the invention.
  • the multi-channel encoder basically comprises an optional pre-processing unit 110, an optional (linear) combination unit 120, a number of encoders 130, 140, a controller 150 and an optional multiplexor (MUX) unit 160.
  • the number N of encoders is equal to or greater than 2, and includes a first encoder 130 and a second encoder 140, and possibly further encoders.
  • the invention considers a multi-channel or polyphonic signal.
  • the initial multi-channel input signal can be provided from an audio signal storage (not shown) or "live", e.g. from a set of microphones (not shown).
  • the audio signals are normally digitized, if not already in digital form, before entering the multi-channel encoder.
  • the multi-channel signal may be provided to the optional pre-processing unit 110 as well as an optional signal combination unit 120 for generating a number N of signal representations, such as for example a main signal representation and an auxiliary signal representation, and possibly further signal representations.
  • the multi-channel or polyphonic signal may be provided to the optional preprocessing unit 110, where different signal conditioning procedures may be performed.
  • the (optionally pre-processed) signals may be provided to an optional signal combination unit 120, which includes a number of combination modules for performing different signal combination procedures, such as linear combinations of the input signals to produce at least a first signal and a second signal.
  • the first encoding process may be a main encoding process and the first signal representation may be a main signal representation.
  • the second encoding process may for example be an auxiliary (side) signal process, and the second signal representation may then be an auxiliary (side) signal representation such as a stereo side signal.
  • traditional stereo coding for example, the L and R channels are summed, and the sum signal is divided by a factor of two in order to provide a traditional mono signal as the first (main) signal.
  • the L and R channels may also be subtracted, and the difference signal is divided by a factor of two to provide a traditional side signal as the second signal.
  • any type of linear combination, or any other type of signal combination for that matter may be performed in the signal combination unit with weighted contributions from at least part of the various channels.
  • the signal combination used by the invention is not limited to two channels but may of course involve multiple channels. It is also possible to generate more than two signals, as indicated in Fig. 5. It is even possible to use one of the input channels directly as a first signal, and another one of the input channels directly as a second signal. For stereo coding, for example, this means that the L channel may be used as main signal and the R channel may be used as side signal, or vice versa.
  • a multitude of other variations also exist.
  • a first signal representation is provided to the first encoder 130, which encodes the first signal according to any suitable encoding principle.
  • a second signal representation is provided to the second encoder 140 for encoding the second signal. If more than two encoders are used, each additional signal representation is normally encoded in a respective encoder.
  • the first encoder may be a main encoder
  • the second encoder may be a side encoder
  • the second side encoder 140 may for example include an adaptive inter-channel prediction (ICP) stage for generating signal reconstruction data based on the first signal representation and the second signal representation.
  • ICP adaptive inter-channel prediction
  • the first (main) signal representation may equivalently be deduced from the signal encoding parameters generated by the first encoder 130, as indicated by the dashed line from the first encoder.
  • the overall multi-channel encoder also comprises a controller 150, which is configured to control a filter smoothing procedure in the second encoder 140 and/or in any of the additional encoders in a signal-adaptive manner in response to characteristics of the multi-channel audio signal.
  • a controller 150 By making the filter smoothing dependent on characteristics of the multi-channel audio signal, such as inter-channel correlation characteristics, it is for example possible to let the controller 150 estimate the expected performance of the encoding process(es) based on the multi-channel audio signal and then adjust the degree and/or type of smoothing accordingly. This will provide a more flexible control so that filter smoothing is performed only when really needed. The better performance, the lesser degree of smoothing is required. The other way around, the worse expected performance of the encoding process, the more smoothing should be applied.
  • the control system which may be realized as a separate controller 150 or integrated in the considered encoder, gives the appropriate control commands to the encoder.
  • the output signals of the various encoders are preferably multiplexed into a single transmission (or storage) signal in the multiplexer unit 160. However, alternatively, the output signals may be transmitted (or stored) separately.
  • encoding is typically performed on a frame-by-frame basis, one frame at a time, and each frame normally comprises audio samples within a pre-defined time period.
  • Fig. 6 is a schematic flow diagram setting forth a basic multi-channel encoding procedure according to a preferred embodiment of the invention.
  • step Sl a first signal representation of one or more audio channels is encoded in a first encoding process.
  • step S2 a second signal representation of one or more audio channels is encoded in a second encoding process.
  • step S3 filter smoothing is performed in the second encoding process or a corresponding decoding process in a signal-adaptive manner, in response to characteristics of the multi-channel audio signal.
  • Fig. 7 is a more detailed schematic flow diagram illustrating an exemplary encoding procedure according to a preferred embodiment of the invention.
  • the first signal representation is encoded in the first encoding process.
  • expected performance of the first encoding process and/or the second encoding process is estimated based on the multi-channel audio input signal.
  • the filter smoothing in the second encoding process is dynamically configured based on the estimated performance. Alternatively, filter smoothing information may be transmitted to the decoding side, in step S 14, as will be explained below.
  • the second signal representation is encoded in the second encoding process, preferably based on the adaptively configured filter smoothing (unless the filter smoothing should be performed on the decoding side).
  • the overall decoding process is generally quite straight forward and basically involves reading the incoming data stream, (possibly interpreting data using transmitted control information), inverse quantization and final reconstruction of the multi-channel audio signal. More specifically, in response to first signal reconstruction data, an encoded first signal representation of at least one of said multiple channels is decoded in a first decoding process. In response to second signal reconstruction data, an encoded second signal representation of at least one of said multiple channels is decoded in a second decoding process.
  • filter smoothing should be performed on the decoding side instead of on the encoding side, information representative of signal-adaptive filter smoothing will have to be transmitted from the encoding side (S 14 in Fig. 7). This enables the decoder to perform signal-adaptive filter smoothing in a corresponding second decoding process based on this information.
  • stereophonic (two-channel) encoding and decoding are generally applicable to multiple channels. Examples include but are not limited to encoding/decoding 5.1 (front left, front centre, front right, rear left and rear right and subwoofer) or 2.1 (left, right and center subwoofer) multi-channel sound.
  • Fig. 8 is a schematic block diagram illustrating relevant parts of an encoder according to an exemplary preferred embodiment of the invention.
  • the encoder basically comprises a first (main) encoder 130 for encoding a first (main) signal such as a typical mono signal, a second (auxiliary/side) encoder 140 for (auxiliary/side) signal encoding, a controller 150 and an optional multiplexor unit 160.
  • the controller 150 is adapted to receive the main signal representation and the side signal representation (or any other appropriate representations of the multi-channel audio signal) and configured to perform the necessary computations to provide adaptive control of the filter smoothing within the side encoder 140.
  • the controller 150 may be a "separate" controller or integrated into the side encoder 140.
  • the encoding parameters are preferably multiplexed into a single transmission or storage signal in the multiplexor unit 160. If filter smoothing is to be performed on the decoding side, the controller generates the appropriate smoothing information and the information is preferably sent to the decoding side via the multiplexor.
  • Fig. 9 is a schematic block diagram illustrating relevant parts of a side encoder and an associated control system according to an exemplary embodiment of the invention.
  • the control system 150 includes a module for estimation of filter performance 152 and a module for filter smoothing configuration.
  • the module 152 for estimation of filter performance preferably operates based on a main signal representation and a side signal representation of the multi-channel audio signal, and estimates the expected performance of a filter in the side encoder 140.
  • the filter may for example be a parametric filter, such as an ICP filter, or any other suitable conventional filter known to the art.
  • the performance may be calculated based on a prediction error. This may equivalently be expressed as a prediction gain.
  • the module 154 for filter smoothing configuration makes the necessary adaptation of the filter smoothing settings in response to the estimated filter performance, and controls the filter smoothing in the side encoder accordingly.
  • Fig. 10 is a schematic block diagram illustrating relevant parts of a decoder according to an exemplary preferred embodiment of the invention.
  • the decoder basically comprises an optional demultiplexor unit 210, a first (main) decoder 230, a second (auxiliary/side) decoder 240, a controller 250, an optional signal combination unit 260 and an optional post-processing unit 270.
  • the demultiplexor 210 preferably separates the incoming reconstruction information such as first (main) signal reconstruction data, second (auxiliary/side) signal reconstruction data and control information such as information on frame division configuration and filter lengths.
  • the first (main) decoder 230 "reconstructs" the first (main) signal in response to the first (main) signal reconstruction data, usually provided in the form of first (main) signal representing encoding parameters.
  • the second (auxiliary/side) decoder 240 preferably "reconstructs" the second (side) signal in response to quantized filter coefficients and the reconstructed first signal representation.
  • the second (side) decoder 240 is also controlled by the controller 250, which may or may not be integrated into the side decoder. In this example, the controller 250 receives smoothing information such as a smoothing factor from the encoding side, and controls the side decoder 240 accordingly.
  • inter-channel prediction (ICP) techniques utilize the inherent inter-channel correlation between the channels.
  • channels are usually represented by the left and the right signals l(n), r(n), an equivalent representation is the mono signal m(n) (a special case of the main signal) and the side signal s(n). Both representations are equivalent and are normally related by the traditional matrix operation:
  • the ICP technique aims to represent the side signal s(n) by an estimate s ⁇ ⁇ ) , which is obtained by filtering the mono signal m(n) through a time-varying FIR filter H(z) having N filter coefficients h t (i):
  • N-I s(n) ⁇ h t (i)m(n ⁇ i) (2)
  • the ICP filter derived at the encoder may for example be estimated by minimizing the mean squared error (MSE), or a related performance measure, for instance psycho- acoustically weighted mean square error, of the side signal prediction error e(n).
  • MSE mean squared error
  • the MSE is typically given by:
  • the optimal filter coefficients are found by minimizing the MSE of the prediction error over all samples and are given by:
  • P ss is the power of the side signal, also expressed as s ⁇ s.
  • the sought filter vector h can now be calculated iteratively in the same way as (10):
  • the optimal ICP (FIR) filter coefficients h opt may be estimated, quantized and sent to the decoder on a frame-by- frame basis.
  • the filter coefficients are treated as vectors, which are efficiently quantized using vector quantization (VQ).
  • VQ vector quantization
  • the quantization of the filter coefficients is one of the most important aspects of the ICP coding procedure.
  • the quantization noise introduced on the filter coefficients can be directly related to the loss in MSE.
  • the target may not always be to minimize the MSE alone but to combine it with smoothing and regularization in order to be able to cope with the cases where there is no correlation between the mono and the side signal.
  • the stereo width i.e. the side signal energy
  • the stereo width is therefore intentionally reduced whenever a problematic frame is encountered.
  • the worst-case scenario i.e. no ICP filtering at all
  • the resulting stereo signal is reduced to pure mono.
  • the frame is not problematic at all, the signal energy does not have to be reduced.
  • the expected filtering performance such as expected prediction gain from the covariance matrix R and the correlation vector r , without having to perform the actual filtering. This is preferably done by a control system as previously described.
  • coding artifacts are mainly present in the reconstructed side signal when the anticipated prediction gain is low or equivalently when the correlation between the mono and the side signal is low.
  • a frame classification algorithm is constructed, which performs classification based on estimated level of prediction gain. For example, when the prediction gain (or the correlation) falls below a certain threshold, the covariance matrix used to derive the ICP filter can be modified according to:
  • the value of the smoothing factor p can be made adaptive to facilitate different levels of modification.
  • the energy of the ICP filter is reduced, thus reducing the energy of the reconstructed side signal.
  • Other schemes for reducing the introduced estimation errors are also plausible. This provides a smoothing effect since the reduction in signal energy generally reduces the differences between different frames, considering the fact that there may originally be large differences in the predicted signal from frame to frame.
  • overlappning windows solves the alising problem for ICP filtering as well.
  • the use of overlapping windows in BCC is not representative of signal- adaptive filter smoothing since there will be a "fixed” smoothing effect and energy reduction for all considered frames irrespective of whether such as reduction is really needed. This results in a rather large performance reduction.
  • a modified cost function is suggested. It is defined as:
  • the smoothing factor ⁇ determines the contribution of the previous ICP filter, thereby controlling the level of smoothing.
  • the proposed filter smoothing effectively removes coding artifacts and stabilizes the stereo image.
  • the problem of stereo image width reduction due to smoothing can be alleviated by making the smoothing factor signal- adaptive, and dependent on the filter performance.
  • a large smoothing factor is preferably used when the prediction gain of the previous filter applied to the current frame is high. However, if the previous filter leads to deterioration in the prediction gain, then the smoothing factor may be gradually decreased.
  • smoothing information such as the smoothing factors described above can be sent to the decoding side, and the signal-adaptive filter smoothing can equivalently be performed on the decoding side rather than on the encoding side.

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Abstract

Une première représentation de signal d'un ou de plusieurs des canaux multiples est codée (S1) dans un premier processus de codage et une seconde représentation de signal d'un ou de plusieurs canaux multiples est codée (S2) dans un second processus de codage par filtre. Le lissage par filtre peut être utilisé pour réduire les effets d'artefacts de codage. Cependant, un lissage par filtre classique conduit généralement à une réduction relativement élevée du rendement et par conséquent n'est pas utilisé largement. Il a été reconnu que les artefacts de codage sont perçus comme étant plus gênants que la réduction temporaire de la largeur stéréophonique et qu'ils sont particulièrement gênants lorsque le filtre de codage fournit une mauvaise estimation du signal cible, les artefacts étant d'autant plus gênants que l'estimation est mauvaise. Par conséquent, un lissage par filtre adaptatif en signal (S3) est introduit dans le second processus de codage ou un processus de décodage correspondant est introduit comme nouveau concept général permettant de résoudre les problèmes de l'état de la technique.
PCT/SE2006/000234 2005-02-23 2006-02-22 Lissage par filtre ameliore dans le codage et/ou le decodage audio multicanaux WO2006091150A1 (fr)

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ES06716924T ES2389499T3 (es) 2005-02-23 2006-02-22 Alisamiento mejorado de filtros en codificación y/o descodificación de audio multi-canal
CN2006800056513A CN101128867B (zh) 2005-02-23 2006-02-22 多声道音频编码和/或解码中改进的滤波器平滑
EP06716924A EP1851759B1 (fr) 2005-02-23 2006-02-22 Lissage par filtre ameliore dans le codage et/ou le decodage audio multicanaux

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SEPCT/SE2005/002033 2005-12-22
PCT/SE2005/002033 WO2006091139A1 (fr) 2005-02-23 2005-12-22 Attribution adaptative de bits pour le codage audio a canaux multiples

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Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2009038512A1 (fr) * 2007-09-19 2009-03-26 Telefonaktiebolaget Lm Ericsson (Publ) Renforcement de réunion d'audio à plusieurs canaux
WO2010128386A1 (fr) * 2009-05-08 2010-11-11 Nokia Corporation Traitement audio multicanaux
WO2012105885A1 (fr) * 2011-02-02 2012-08-09 Telefonaktiebolaget L M Ericsson (Publ) Détermination de la différence de temps entre canaux pour un signal audio multicanal
WO2013149672A1 (fr) * 2012-04-05 2013-10-10 Huawei Technologies Co., Ltd. Procédé de détermination d'un paramètre de codage pour un signal audio multicanal et codeur audio multicanal
US9082395B2 (en) 2009-03-17 2015-07-14 Dolby International Ab Advanced stereo coding based on a combination of adaptively selectable left/right or mid/side stereo coding and of parametric stereo coding
US10586545B2 (en) 2010-04-09 2020-03-10 Dolby International Ab MDCT-based complex prediction stereo coding

Families Citing this family (19)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
BRPI0305434B1 (pt) * 2002-07-12 2017-06-27 Koninklijke Philips Electronics N.V. Methods and arrangements for encoding and decoding a multichannel audio signal, and multichannel audio coded signal
US8090573B2 (en) * 2006-01-20 2012-01-03 Qualcomm Incorporated Selection of encoding modes and/or encoding rates for speech compression with open loop re-decision
US8346544B2 (en) * 2006-01-20 2013-01-01 Qualcomm Incorporated Selection of encoding modes and/or encoding rates for speech compression with closed loop re-decision
US8032369B2 (en) * 2006-01-20 2011-10-04 Qualcomm Incorporated Arbitrary average data rates for variable rate coders
KR20080053739A (ko) * 2006-12-11 2008-06-16 삼성전자주식회사 적응적으로 윈도우 크기를 적용하는 부호화 장치 및 방법
WO2008084427A2 (fr) * 2007-01-10 2008-07-17 Koninklijke Philips Electronics N.V. Décodeur audio
WO2008126382A1 (fr) * 2007-03-30 2008-10-23 Panasonic Corporation Dispositif et procédé de codage
ES2401817T3 (es) * 2008-01-31 2013-04-24 Agency For Science, Technology And Research Procedimiento y dispositivo de distribución/truncado de la velocidad de transmisión de bits para codificación de audio escalable
CN102016982B (zh) 2009-02-04 2014-08-27 松下电器产业株式会社 结合装置、远程通信系统以及结合方法
EP2434483A4 (fr) * 2009-05-20 2016-04-27 Panasonic Ip Corp America Dispositif d'encodage, dispositif de décodage et procédés associés
US8700410B2 (en) * 2009-06-18 2014-04-15 Texas Instruments Incorporated Method and system for lossless value-location encoding
KR101613975B1 (ko) 2009-08-18 2016-05-02 삼성전자주식회사 멀티 채널 오디오 신호의 부호화 방법 및 장치, 그 복호화 방법 및 장치
WO2014046916A1 (fr) * 2012-09-21 2014-03-27 Dolby Laboratories Licensing Corporation Approche de codage audio spatial en couches
ES2955962T3 (es) 2015-09-25 2023-12-11 Voiceage Corp Método y sistema que utiliza una diferencia de correlación a largo plazo entre los canales izquierdo y derecho para mezcla descendente en el dominio del tiempo de una señal de sonido estéreo en canales primarios y secundarios
CA3011915C (fr) 2016-01-22 2021-07-13 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Appareil et procede pour estimer une difference de temps inter-canaux
CN111133510B (zh) 2017-09-20 2023-08-22 沃伊斯亚吉公司 用于在celp编解码器中高效地分配比特预算的方法和设备
US10553224B2 (en) * 2017-10-03 2020-02-04 Dolby Laboratories Licensing Corporation Method and system for inter-channel coding
US20200402521A1 (en) * 2019-06-24 2020-12-24 Qualcomm Incorporated Performing psychoacoustic audio coding based on operating conditions
CA3145047A1 (fr) * 2019-07-08 2021-01-14 Voiceage Corporation Procede et systeme permettant de coder des metadonnees dans des flux audio et permettant une attribution de debit binaire efficace a des flux audio codant

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20060004583A1 (en) * 2004-06-30 2006-01-05 Juergen Herre Multi-channel synthesizer and method for generating a multi-channel output signal

Family Cites Families (43)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5434948A (en) * 1989-06-15 1995-07-18 British Telecommunications Public Limited Company Polyphonic coding
CN1062963C (zh) * 1990-04-12 2001-03-07 多尔拜实验特许公司 用于产生高质量声音信号的解码器和编码器
NL9100173A (nl) 1991-02-01 1992-09-01 Philips Nv Subbandkodeerinrichting, en een zender voorzien van de kodeerinrichting.
US5285498A (en) 1992-03-02 1994-02-08 At&T Bell Laboratories Method and apparatus for coding audio signals based on perceptual model
US5694332A (en) * 1994-12-13 1997-12-02 Lsi Logic Corporation MPEG audio decoding system with subframe input buffering
US5695332A (en) * 1995-11-23 1997-12-09 Samuels; Russell H. A. Orthodontic facebow with locking catch
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
US5812971A (en) * 1996-03-22 1998-09-22 Lucent Technologies Inc. Enhanced joint stereo coding method using temporal envelope shaping
US5796842A (en) 1996-06-07 1998-08-18 That Corporation BTSC encoder
DE19628293C1 (de) * 1996-07-12 1997-12-11 Fraunhofer Ges Forschung Codieren und Decodieren von Audiosignalen unter Verwendung von Intensity-Stereo und Prädiktion
SE9700772D0 (sv) 1997-03-03 1997-03-03 Ericsson Telefon Ab L M A high resolution post processing method for a speech decoder
JPH1132399A (ja) 1997-05-13 1999-02-02 Sony Corp 符号化方法及び装置、並びに記録媒体
US6012031A (en) * 1997-09-24 2000-01-04 Sony Corporation Variable-length moving-average filter
US6591241B1 (en) * 1997-12-27 2003-07-08 Stmicroelectronics Asia Pacific Pte Limited Selecting a coupling scheme for each subband for estimation of coupling parameters in a transform coder for high quality audio
JP3606458B2 (ja) 1998-10-13 2005-01-05 日本ビクター株式会社 音声信号伝送方法及び音声復号方法
US6446037B1 (en) * 1999-08-09 2002-09-03 Dolby Laboratories Licensing Corporation Scalable coding method for high quality audio
JP2001184090A (ja) 1999-12-27 2001-07-06 Fuji Techno Enterprise:Kk 信号符号化装置，及び信号復号化装置，並びに信号符号化プログラムを記録したコンピュータ読み取り可能な記録媒体，及び信号復号化プログラムを記録したコンピュータ読み取り可能な記録媒体
JP3335605B2 (ja) 2000-03-13 2002-10-21 日本電信電話株式会社 ステレオ信号符号化方法
SE519985C2 (sv) 2000-09-15 2003-05-06 Ericsson Telefon Ab L M Kodning och avkodning av signaler från flera kanaler
SE519981C2 (sv) 2000-09-15 2003-05-06 Ericsson Telefon Ab L M Kodning och avkodning av signaler från flera kanaler
JP3894722B2 (ja) 2000-10-27 2007-03-22 松下電器産業株式会社 ステレオオーディオ信号高能率符号化装置
JP3846194B2 (ja) 2001-01-18 2006-11-15 日本ビクター株式会社 音声符号化方法，音声復号化方法，音声受信装置及び音声信号伝送方法
DE60209888T2 (de) * 2001-05-08 2006-11-23 Koninklijke Philips Electronics N.V. Kodieren eines audiosignals
SE0202159D0 (sv) * 2001-07-10 2002-07-09 Coding Technologies Sweden Ab Efficientand scalable parametric stereo coding for low bitrate applications
US7240001B2 (en) * 2001-12-14 2007-07-03 Microsoft Corporation Quality improvement techniques in an audio encoder
US7460993B2 (en) * 2001-12-14 2008-12-02 Microsoft Corporation Adaptive window-size selection in transform coding
KR101016251B1 (ko) * 2002-04-10 2011-02-25 코닌클리케 필립스 일렉트로닉스 엔.브이. 스테레오 신호의 코딩
JP4714416B2 (ja) 2002-04-22 2011-06-29 コーニンクレッカ フィリップス エレクトロニクス エヌ ヴィ 空間的オーディオのパラメータ表示
DE60311794C5 (de) 2002-04-22 2022-11-10 Koninklijke Philips N.V. Signalsynthese
EP1500083B1 (fr) 2002-04-22 2006-06-28 Koninklijke Philips Electronics N.V. Representation parametrique de signaux audio multicanaux
JP4062971B2 (ja) 2002-05-27 2008-03-19 松下電器産業株式会社 オーディオ信号符号化方法
US7099513B2 (en) * 2002-06-18 2006-08-29 Pts Corporation Bit allocation process for multi-stage image compression
BRPI0305434B1 (pt) * 2002-07-12 2017-06-27 Koninklijke Philips Electronics N.V. Methods and arrangements for encoding and decoding a multichannel audio signal, and multichannel audio coded signal
JP4022111B2 (ja) 2002-08-23 2007-12-12 株式会社エヌ・ティ・ティ・ドコモ 信号符号化装置及び信号符号化方法
JP4373693B2 (ja) 2003-03-28 2009-11-25 パナソニック株式会社 音響信号の階層符号化方法および階層復号化方法
AU2003222397A1 (en) * 2003-04-30 2004-11-23 Nokia Corporation Support of a multichannel audio extension
DE10328777A1 (de) 2003-06-25 2005-01-27 Coding Technologies Ab Vorrichtung und Verfahren zum Codieren eines Audiosignals und Vorrichtung und Verfahren zum Decodieren eines codierten Audiosignals
US7809579B2 (en) * 2003-12-19 2010-10-05 Telefonaktiebolaget Lm Ericsson (Publ) Fidelity-optimized variable frame length encoding
US7356748B2 (en) * 2003-12-19 2008-04-08 Telefonaktiebolaget Lm Ericsson (Publ) Partial spectral loss concealment in transform codecs
US7725324B2 (en) * 2003-12-19 2010-05-25 Telefonaktiebolaget Lm Ericsson (Publ) Constrained filter encoding of polyphonic signals
US7460990B2 (en) * 2004-01-23 2008-12-02 Microsoft Corporation Efficient coding of digital media spectral data using wide-sense perceptual similarity
DE102004009954B4 (de) * 2004-03-01 2005-12-15 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Verarbeiten eines Multikanalsignals
EP1851866B1 (fr) * 2005-02-23 2011-08-17 Telefonaktiebolaget LM Ericsson (publ) Attribution adaptative de bits pour le codage audio a canaux multiples

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20060004583A1 (en) * 2004-06-30 2006-01-05 Juergen Herre Multi-channel synthesizer and method for generating a multi-channel output signal

Non-Patent Citations (4)

* Cited by examiner, † Cited by third party
Title
DATABASE INSPEC [online] Database accession no. (7424360) *
PURNHAGEN H.: "Low complexity parametric stereo coding in MPEG-4", PROC. OF THE 7TH INT. CONFERENCE ON DIGITAL AUDIO EFFECTS (DAFX-04), 5 October 2004 (2004-10-05) - 8 October 2004 (2004-10-08), NAPLES, IT, pages 163 - 168, XP002364489, Retrieved from the Internet <URL:http://www.dafx04.na.infn.it/WebProc/Proc/P_163.pdf> *
SCHULLER G.D.T. ET AL.: "Perceptual audio coding using adaptive pre- and post-filters and lossless compression", SPEECH AND AUDIO PROCESSING, IEEE TRANSACTIONS, vol. 10, no. 6, September 2002 (2002-09-01), pages 379 - 390, XP001219570 *
See also references of EP1851759A4 *

Cited By (28)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR101450940B1 (ko) 2007-09-19 2014-10-15 텔레폰악티에볼라겟엘엠에릭슨(펍) 멀티채널 오디오의 조인트 인핸스먼트
WO2009038512A1 (fr) * 2007-09-19 2009-03-26 Telefonaktiebolaget Lm Ericsson (Publ) Renforcement de réunion d'audio à plusieurs canaux
US8218775B2 (en) 2007-09-19 2012-07-10 Telefonaktiebolaget L M Ericsson (Publ) Joint enhancement of multi-channel audio
US9905230B2 (en) 2009-03-17 2018-02-27 Dolby International Ab Advanced stereo coding based on a combination of adaptively selectable left/right or mid/side stereo coding and of parametric stereo coding
US10297259B2 (en) 2009-03-17 2019-05-21 Dolby International Ab Advanced stereo coding based on a combination of adaptively selectable left/right or mid/side stereo coding and of parametric stereo coding
US11322161B2 (en) 2009-03-17 2022-05-03 Dolby International Ab Audio encoder with selectable L/R or M/S coding
US11315576B2 (en) 2009-03-17 2022-04-26 Dolby International Ab Selectable linear predictive or transform coding modes with advanced stereo coding
US11133013B2 (en) 2009-03-17 2021-09-28 Dolby International Ab Audio encoder with selectable L/R or M/S coding
US9082395B2 (en) 2009-03-17 2015-07-14 Dolby International Ab Advanced stereo coding based on a combination of adaptively selectable left/right or mid/side stereo coding and of parametric stereo coding
US11017785B2 (en) 2009-03-17 2021-05-25 Dolby International Ab Advanced stereo coding based on a combination of adaptively selectable left/right or mid/side stereo coding and of parametric stereo coding
TWI508058B (zh) * 2009-05-08 2015-11-11 Nokia Corp 多聲道音訊處理技術
WO2010128386A1 (fr) * 2009-05-08 2010-11-11 Nokia Corporation Traitement audio multicanaux
US9129593B2 (en) 2009-05-08 2015-09-08 Nokia Technologies Oy Multi channel audio processing
US11217259B2 (en) 2010-04-09 2022-01-04 Dolby International Ab Audio upmixer operable in prediction or non-prediction mode
RU2717387C1 (ru) * 2010-04-09 2020-03-23 Долби Интернешнл Аб Устройство повышающего микширования звука, выполненное с возможностью работы в режиме с предсказанием или в режиме без предсказания
US11810582B2 (en) 2010-04-09 2023-11-07 Dolby International Ab MDCT-based complex prediction stereo coding
US11264038B2 (en) 2010-04-09 2022-03-01 Dolby International Ab MDCT-based complex prediction stereo coding
US10734002B2 (en) 2010-04-09 2020-08-04 Dolby International Ab Audio upmixer operable in prediction or non-prediction mode
US10586545B2 (en) 2010-04-09 2020-03-10 Dolby International Ab MDCT-based complex prediction stereo coding
US10573328B2 (en) 2011-02-02 2020-02-25 Telefonaktiebolaget Lm Ericsson (Publ) Determining the inter-channel time difference of a multi-channel audio signal
US9525956B2 (en) 2011-02-02 2016-12-20 Telefonaktiebolaget Lm Ericsson (Publ) Determining the inter-channel time difference of a multi-channel audio signal
WO2012105885A1 (fr) * 2011-02-02 2012-08-09 Telefonaktiebolaget L M Ericsson (Publ) Détermination de la différence de temps entre canaux pour un signal audio multicanal
US9424852B2 (en) 2011-02-02 2016-08-23 Telefonaktiebolaget Lm Ericsson (Publ) Determining the inter-channel time difference of a multi-channel audio signal
US10332529B2 (en) 2011-02-02 2019-06-25 Telefonaktiebolaget Lm Ericsson (Publ) Determining the inter-channel time difference of a multi-channel audio signal
CN103403800A (zh) * 2011-02-02 2013-11-20 瑞典爱立信有限公司 确定多声道音频信号的声道间时间差
US9449604B2 (en) 2012-04-05 2016-09-20 Huawei Technologies Co., Ltd. Method for determining an encoding parameter for a multi-channel audio signal and multi-channel audio encoder
CN103460283A (zh) * 2012-04-05 2013-12-18 华为技术有限公司 确定多信道音频信号的编码参数的方法及多信道音频编码器
WO2013149672A1 (fr) * 2012-04-05 2013-10-10 Huawei Technologies Co., Ltd. Procédé de détermination d'un paramètre de codage pour un signal audio multicanal et codeur audio multicanal

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EP1856688B1 (fr) 2011-07-27
WO2006091150B1 (fr) 2006-12-14
US9626973B2 (en) 2017-04-18
EP1856688A4 (fr) 2010-07-28
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EP1856688A1 (fr) 2007-11-21
US20080262850A1 (en) 2008-10-23

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