WO2007011079A1 - Appareil et procede de codage et de decodage de signal audio - Google Patents

Appareil et procede de codage et de decodage de signal audio Download PDF

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Publication number
WO2007011079A1
WO2007011079A1 PCT/KR2005/002291 KR2005002291W WO2007011079A1 WO 2007011079 A1 WO2007011079 A1 WO 2007011079A1 KR 2005002291 W KR2005002291 W KR 2005002291W WO 2007011079 A1 WO2007011079 A1 WO 2007011079A1
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WO
WIPO (PCT)
Prior art keywords
block
channels
subdivided
blocks
switching
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PCT/KR2005/002291
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English (en)
Inventor
Tilman Liebchen
Original Assignee
Lg Electronics Inc.
Noll, Peter
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Publication date
Application filed by Lg Electronics Inc., Noll, Peter filed Critical Lg Electronics Inc.
Priority to PCT/KR2005/002291 priority Critical patent/WO2007011079A1/fr
Priority to US11/481,933 priority patent/US7966190B2/en
Priority to US11/481,916 priority patent/US8108219B2/en
Priority to US11/481,942 priority patent/US7830921B2/en
Priority to US11/481,926 priority patent/US7949014B2/en
Priority to US11/481,917 priority patent/US7991272B2/en
Priority to US11/481,929 priority patent/US7991012B2/en
Priority to US11/481,930 priority patent/US8032368B2/en
Priority to US11/481,941 priority patent/US8050915B2/en
Priority to US11/481,931 priority patent/US7411528B2/en
Priority to US11/481,939 priority patent/US8121836B2/en
Priority to US11/481,940 priority patent/US8180631B2/en
Priority to US11/481,915 priority patent/US7996216B2/en
Priority to US11/481,927 priority patent/US7835917B2/en
Priority to US11/481,932 priority patent/US8032240B2/en
Priority to JP2008521306A priority patent/JP2009500682A/ja
Priority to JP2008521310A priority patent/JP2009500686A/ja
Priority to JP2008521319A priority patent/JP2009500693A/ja
Priority to PCT/KR2006/002690 priority patent/WO2007008012A2/fr
Priority to CN2006800294070A priority patent/CN101243496B/zh
Priority to PCT/KR2006/002689 priority patent/WO2007008011A2/fr
Priority to PCT/KR2006/002691 priority patent/WO2007008013A2/fr
Priority to CNA2006800289829A priority patent/CN101238510A/zh
Priority to EP06769219A priority patent/EP1913584A4/fr
Priority to PCT/KR2006/002678 priority patent/WO2007008000A2/fr
Priority to JP2008521308A priority patent/JP2009500684A/ja
Priority to EP06769224A priority patent/EP1913794A4/fr
Priority to CNA2006800294174A priority patent/CN101243489A/zh
Priority to JP2008521305A priority patent/JP2009500681A/ja
Priority to EP06769218A priority patent/EP1913589A4/fr
Priority to CNA2006800304693A priority patent/CN101243492A/zh
Priority to EP06769220A priority patent/EP1913585A4/fr
Priority to CNA2006800305412A priority patent/CN101243497A/zh
Priority to EP06769223A priority patent/EP1913587A4/fr
Priority to CNA2006800251376A priority patent/CN101218631A/zh
Priority to PCT/KR2006/002677 priority patent/WO2007007999A2/fr
Priority to JP2008521307A priority patent/JP2009500683A/ja
Priority to EP06769226A priority patent/EP1913588A4/fr
Priority to EP06757764A priority patent/EP1913579A4/fr
Priority to PCT/KR2006/002682 priority patent/WO2007008004A2/fr
Priority to PCT/KR2006/002687 priority patent/WO2007008009A1/fr
Priority to CNA2006800251395A priority patent/CN101218629A/zh
Priority to JP2008521318A priority patent/JP2009500692A/ja
Priority to EP06769225A priority patent/EP1911021A4/fr
Priority to PCT/KR2006/002680 priority patent/WO2007008002A2/fr
Priority to JP2008521314A priority patent/JP2009500689A/ja
Priority to PCT/KR2006/002679 priority patent/WO2007008001A2/fr
Priority to PCT/KR2006/002686 priority patent/WO2007008008A2/fr
Priority to EP06757765A priority patent/EP1913580A4/fr
Priority to EP06769227A priority patent/EP1911020A4/fr
Priority to CNA2006800305111A priority patent/CN101243494A/zh
Priority to CNA200680024866XA priority patent/CN101218852A/zh
Priority to PCT/KR2006/002681 priority patent/WO2007008003A2/fr
Priority to EP06757767A priority patent/EP1913582A4/fr
Priority to EP06757768A priority patent/EP1913583A4/fr
Priority to JP2008521313A priority patent/JP2009500688A/ja
Priority to CNA2006800304797A priority patent/CN101243493A/zh
Priority to JP2008521309A priority patent/JP2009500685A/ja
Priority to PCT/KR2006/002685 priority patent/WO2007008007A1/fr
Priority to JP2008521315A priority patent/JP2009500690A/ja
Priority to CN2006800251380A priority patent/CN101218628B/zh
Priority to EP06769222A priority patent/EP1908058A4/fr
Priority to JP2008521311A priority patent/JP2009500687A/ja
Priority to CNA2006800305499A priority patent/CN101243495A/zh
Priority to PCT/KR2006/002683 priority patent/WO2007008005A1/fr
Priority to CNA200680028892XA priority patent/CN101238509A/zh
Priority to PCT/KR2006/002688 priority patent/WO2007008010A1/fr
Priority to EP06757766A priority patent/EP1913581A4/fr
Priority to CN2006800252699A priority patent/CN101218630B/zh
Priority to JP2008521317A priority patent/JP2009500691A/ja
Priority to JP2008521316A priority patent/JP2009510810A/ja
Publication of WO2007011079A1 publication Critical patent/WO2007011079A1/fr
Priority to US12/232,527 priority patent/US7962332B2/en
Priority to US12/232,526 priority patent/US8010372B2/en
Priority to US12/232,590 priority patent/US8055507B2/en
Priority to US12/232,593 priority patent/US8326132B2/en
Priority to US12/232,591 priority patent/US8255227B2/en
Priority to US12/232,595 priority patent/US8417100B2/en
Priority to US12/232,662 priority patent/US8510120B2/en
Priority to US12/232,659 priority patent/US8554568B2/en
Priority to US12/232,658 priority patent/US8510119B2/en
Priority to US12/232,744 priority patent/US8032386B2/en
Priority to US12/232,739 priority patent/US8155152B2/en
Priority to US12/232,741 priority patent/US8149877B2/en
Priority to US12/232,740 priority patent/US8149876B2/en
Priority to US12/232,748 priority patent/US8155153B2/en
Priority to US12/232,747 priority patent/US8149878B2/en
Priority to US12/232,743 priority patent/US7987008B2/en
Priority to US12/232,734 priority patent/US8155144B2/en
Priority to US12/232,781 priority patent/US7930177B2/en
Priority to US12/232,782 priority patent/US8046092B2/en
Priority to US12/232,784 priority patent/US7987009B2/en
Priority to US12/232,783 priority patent/US8275476B2/en
Priority to US12/314,891 priority patent/US8065158B2/en

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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
    • G10L19/02—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
    • G10L19/022—Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring

Definitions

  • the present invention relates to a method for processing audio signal, and more particularly to a method and apparatus of encoding and decoding audio signal.
  • Lossless reconstruction is becoming a more important feature than high efficiency in compression by means of perceptual coding as defined in MPEG standards such as MP3 or AAC.
  • MPEG standards such as MP3 or AAC.
  • DVD audio and Super CD Audio include proprietary lossless compression schemes
  • a new lossless coding scheme has been considered as an extension to the MPEG-4 Audio standard. Lossless audio coding permits the compression of digital audio data without any loss in quality due to a perfect reconstruction of the original signal.
  • the present invention relates to a method for processing forward- adaptive linear prediction, which offers remarkable compression even with low predictor orders. Nevertheless, performance can be significantly improved by using higher predictor orders, more efficient quantization and encoding of the predictor coefficients, and adaptive block length switching.
  • Audio Lossless Coding will define methods for lossless coding of audio signals with arbitrary sampling rates, resolutions of up to 32 bit, and up to 256 channels.
  • the lossless codec uses forward-adaptive Linear Predictive Coding (LPC) to reduce bit rates compared to PCM, leaving the optimization entirely to the encoder.
  • LPC Linear Predictive Coding
  • various encoder implementations are possible, offering a certain range in terms of efficiency and complexity. Although remarkable compression is achieved even for low predictor orders, still better compression becomes possible using high-order prediction. In this case, more efficient coding of the predictor coefficients is necessary in order to limit the amount of side information.
  • the present invention relate to an encoder and/or decoder (including methods of encoding and decoding) data. Data may be encoded or decoded in a lossless manner.
  • Embodiments relate to a flexible, hierarchical block switch scheme, allowing for up to six different block lengths within a frame.
  • Embodiments relate to independent block switching for each channel.
  • Embodiments relate to a maximum predictor order of 1023.
  • a method of processing an audio signal includes the steps of switching a plurality of channels of an audio data frame independently when the channels are not correlated with each other, switching the channels synchronously when the channels are correlated with each other, where each channel is subdivided into a plurality of blocks having non-uniform lengths.
  • the blocks may be hierarchically at one or more block switching levels, and each block may result from a subdivision of a superordinate block of double length.
  • a method of encoding an audio signal includes the steps of switching a plurality of channels of an audio data frame independently when the channels are not correlated with each other, switching the channels synchronously when the channels are correlated with each other, and generating block switching information indicating how a plurality of blocks are subdivided from the channels at one or more block switching levels.
  • a method of decoding an audio signal includes the steps of receiving an audio data frame having a plurality of channels, parsing block switching information from the audio data frame to determine how a plurality of blocks are subdivided from the channels, and identifying and decoding the subdivided blocks using the parsed block switching information.
  • the channels included in the audio data frame are switched independently when the channels are not correlated with each other and the channels are switched synchronously when the channels are correlated with each other.
  • an apparatus of encoding an audio signal includes an encoder which switches a plurality of channels of an audio data frame independently when the channels are not correlated with each other, and it switches the channels synchronously when the channels are correlated with each other. The encoder then generates block switching information indicating how a plurality of blocks are subdivided from the channels at one or more block switching levels, respectively.
  • an apparatus of decoding an audio signal includes a decoder which receives an audio data frame having a plurality of channels. The channels are switched independently when the channels are not correlated with each other and are switched synchronously when they are correlated with each other. The decoder then parses block switching information from the audio data frame for determining how a plurality of blocks are subdivided at one or more block switching levels, and it identifies and decode the subdivided blocks using the parsed block switching information.
  • Figure 1 is an example illustration of an audio signal encoder.
  • Figure 2 is an example illustration of an audio signal decoder.
  • Figure 3 is an measured distributions of parcor coefficients for 48KHz, 16-bit audio material.
  • Figure 4 is an compander functions C(r) and -C(-r).
  • Figure 5 is an example of a block switching hierarchy structure.
  • Figure 6 is an example of a block switching examples and corresponding block switching information codes.
  • Figure 7 is an example of a bit stream of old block switching scheme.
  • Figure 8 is an example of a bit stream of new block switching (BS) scheme: No BS (top), synchronized BS between CPE channels 1 and 2 (middle), independent BS (bottom).
  • BS new block switching
  • Figure 9 is a switched difference coding scheme.
  • Figure 10 is a partition of the residual distribution.
  • Figure 1 shows the typical processing for one input channel of audio data.
  • a buffer stores one block of input samples, and an optimum set of parcor coefficients is calculated for each block.
  • the number of coefficients, i.e. the order of the predictor, can be adaptively chosen as well.
  • the quantized parcor values are entropy coded for transmission, and converted to LPC coefficients for the prediction filter which calculates the prediction residual.
  • the residual is entropy coded using different entropy codes.
  • the indices of the chosen codes have to be transmitted as side information.
  • a multiplexing unit combines coded residual, code indices, predictor coefficients and other additional information to form the compressed bitstream.
  • the encoder also provides a CRC checksum, which is supplied mainly for the decoder to verify the decoded data. On the encoder side, the
  • CRC can be used to ensure that the compressed data is losslessly decodable.
  • Additional encoder options comprise block length switching, random access and joint channel coding.
  • the encoder may use these options to offer several compression levels with different complexities.
  • the basic version of the encoder uses a fixed block length.
  • the encoder can switch between different block lengths to adapt to stationary regions as well as to transient segments of the audio signal.
  • the codec allows random access in defined intervals down to some milliseconds, depending on the block length.
  • joint channel coding is used to exploit dependencies between channels of stereo or multi-channel signals. This can be achieved by coding the difference between two channels in those segments where this difference can be coded more efficiently than one of the original channels.
  • the entropy coding part of the prediction residual provides two alternative coding techniques with different complexities. Besides low complexity yet efficient Golomb-Rice coding, the BGMC arithmetic coding scheme offers even better compression at the expense of a slightly increased complexity.
  • the encoder will also offer efficient compression of floating-point audio data in the 32-bit IEEE format.
  • This codec extension employs an algorithm that basically splits the floating-point signal into a truncated integer signal and a difference signal which contains the remaining fractional part. The integer signal is then compressed using the normal encoding scheme for PCM signals, while the difference signal is coded separately. A detailed description of the floating-point extension can be found.
  • the Figure 2 shows the lossless audio signal decoder which is significantly less complex than the encoder, since no adaptation has to be carried out.
  • the decoder merely decodes the entropy coded residual and the parcor values, converts them into LPC coefficients, and applies the inverse prediction filter to calculate the lossless reconstruction signal.
  • the computational effort of the decoder mainly depends on the predictor orders chosen by the encoder. Since the average order is typically well below the maximum order, prediction with greater maximum orders does not necessarily lead to a significant increase of decoder complexity. In most cases, realtime decoding is possible even on low-end systems.
  • Linear prediction is used in many applications for speech and audio signal processing. In the following, only FIR predictors are considered. Prediction with FIR Filters
  • the current sample of a time-discrete signal x(n) can be
  • the procedure of estimating the predictor coefficients from a segment of input samples, prior to filtering that segment, is referred to as forward adaptation. In that case, the coefficients have to be transmitted. If the coefficients are estimated from previously processed segments or samples, e.g. from the residual, we speak of backward adaptation. This procedure has the advantage that no transmission of the coefficients is needed, since the data required to estimate the coefficients is available to the decoder as well.
  • Forward-adaptive prediction with orders around 10 is widely used in speech coding, and can be employed for lossless audio coding as well.
  • An exception is the special 1-bit lossless codec for the Super Audio CD, which uses predictor orders of up to 128.
  • backward-adaptive FIR filters with some hundred coefficients are commonly used in many areas, e.g. channel equalization and echo cancellation.
  • Most systems are based on the LMS algorithm or a variation thereof, which has also been proposed for lossless audio coding.
  • LMS-based coding schemes with high orders are applicable since the predictor coefficients do not have to be transmitted as side information, thus their number does not contribute to the data rate.
  • backward- adaptive codecs have the drawback that the adaptation has to be carried out both in the encoder and the decoder, making the decoder significantly more complex than in the forward-adaptive case.
  • the optimal predictor coefficients h k (in terms of a minimized variance of the residual) are usually estimated for
  • each block by the autocorrelation method or the covariance method.
  • the autocorrelation method using the Levinson-Durbin algorithm, has the additional advantage of providing a simple means to iteratively adapt the order of the predictor. Furthermore, the algorithm inherently calculates the corresponding parcor coefficients as well.
  • Another crucial point in forward-adaptive prediction is to determine a suitable predictor order. Increasing the order decreases the variance of the
  • the task is to find the optimum order which minimizes the total bit rate. This can be expressed by minimizing
  • the search for the optimum order can be carried out efficiently by the Levinson-Durbin algorithm, which determines recursively all predictors with increasing order. For each order, a complete set of predictor coefficients is
  • the variance ⁇ ] of the corresponding residual can be
  • the total bit rate can be determined in each iteration, i.e. for each predictor order. The optimum order is found at the point where the total bit rate no longer decreases.
  • the first two parcor coefficients r, and r 2 are typically very close to -1 and
  • the direct form predictor filter uses predictor coefficients h k
  • a lossless coding method specifies an integer-arithmetic function for conversion between quantized
  • Embodiments relate to encoders, decoders, methods of encoding, and methods of decoding.
  • an encoder is at least one of an audio encoder, and an Audio Lossless Coding encoder.
  • a method of encoding is implemented in at least one of an audio encoder, and an Audio Lossless Coding encoder.
  • a decoder is at least one of an audio decoder, and an Audio Lossless Coding decoder.
  • a method of decoding is implemented in at least one of an audio decoder, and an Audio Lossless Coding decoder.
  • Embodiments relate to a block switching mechanism which subdivides a frame of audio data into four quarter-length blocks, instead of encoding it as one single block. Switching between one long and four short blocks may be performed adaptively on a frame-by-frame basis.
  • a more flexible, hierarchical block switching scheme allows for up to six different blocks lengths (differing by factors of two) within a frame.
  • independent block switching for each channel may be implemented (e.g. each channel pair may be switched independently in the case of joint channel coding).
  • a maximum predictor order of 1023 may be implemented.
  • Audio Lossless Coding includes a relatively simple block switching mechanism. Each frame of N samples is either encoded using one
  • this scheme may have some limitations. For example, only 1 :4 switching may be possible, although different switching (e.g. 1 :2, 1 :8, and combinations thereof) may be more efficient in some cases. For example, switching is done identically for all channels, although different channels may require different switching (which is especially true if the channels are not correlated).
  • a relatively flexible block switching scheme may be implemented, where each frame can be hierarchically subdivided into many blocks.
  • Figure 5 illustrates a frame which can be hierarchically
  • N/2, N/4, N/8, N/16, and N/32 may be possible within a frame, as long as each block results from a subdivision of a superordinate block of double length, in accordance with embodiments.
  • a partition into N/4 + N/4 + N/2 may be possible, while a partition into N/4 + N/2 + N/4 may not be possible.
  • the actual partition may be signaled in an additional field block switching information(bs_info) (illustrated in the right column of Figure 6), where the length depends on the number of block switching levels.
  • Bs_info block switching information
  • Table 1 Block switching levels.
  • the bs_info field may include up to 4 bytes, in accordance with embodiments.
  • the mapping of bits with respect to the levels 1 to 5 may be [(0)1223333 44444444 55555555 55555555].
  • bs_info are set if a block is further subdivided. For the topmost example there is no subdivision at all, thus the code is (0)0000000.
  • the frame in the second row is subdivided ((0)1%), where only the second block of length N/2 is further split ((0)101%) into two blocks of length N/4. If an N/4 block is split as in the fourth row, it is indicated in the following bits ((0)111 0100).
  • bs_info fields may be transmitted for all channel pairs (CPEs) and all single channels (SCEs), enabling independent block switching for different channels, in accordance with embodiments.
  • bs_info field for each CPE and SCE in a frame (e.g. the two channels of a CPE are switched synchronously), in accordance with embodiments. If they are switched independently, the first bit of bs_info may be set to 1 , and the information applies to the CPE's first channel. In this example, another bs_info field for the second channel becomes necessary.
  • the arrangement of blocks in the bit stream can be dynamically arranged.
  • all channels use the same partition (e.g. either one long or four short blocks) and corresponding short blocks of different channels are arranged successively (e.g. blocks 1.1 , 2.1 , and 3.1 ), leading to an interleaved structure.
  • short blocks are only interleaved if they belong to a channel pair that uses difference coding and therefore synchronized block switching (e.g. the middle row of Figure 8). This interleaving may be beneficial, since in a channel pair a block of one channel
  • channel data can be arranged separately (e.g. bottom row of Figure 8).
  • Embodiments relate to higher predictor orders. Absent hierarchical block switching, there may be a factor of 4 between the long and the short block length (e.g. 4096 & 1024 or 8192 & 2048), in accordance with embodiments. In embodiments (e.g. where hierarchical block switching is implemented), this factor can be increased (e.g. up to 32), enabling a larger range (e.g. 16384 down to 512 or even 32768 to 1024 for high sampling rates).
  • K mm 1023.
  • K ⁇ may be bound by the block length N B ,
  • the max_order field in the file header is 10 bits.
  • the opt_order field of the block data is 10 bits. The actual number of bits in a particular block may depend on the maximum order allowed for a block. If the block is short, this local maximum order may be smaller than the global maximum order (stated in max_order in the file
  • a first sample of a current block is predicted using the last K samples of a previous block.
  • the K value is determined from the opt_order which is derived the aboved equation.
  • the current block is a channel's first block
  • no samples from the previous block may be used.
  • prediction with progressive order is employed, where the scaled parcor coefficients are converted progressively to LPC coefficient inside the prediction filter.
  • Random access stands for fast access to any part of the encoded audio signal without costly decoding of previous parts. It is an important feature for applications that employ seeking, editing, or streaming of the compressed data.
  • the encoder In order to enable random access, the encoder has to insert frames that can be decoded without decoding previous frames. In those random access frames, no samples from previous frames may be used for prediction.
  • the distance between random access frames can be chosen from 255 to one frame. Depending on frame length and sampling rate, random access down to some milliseconds is possible.
  • a conventional K-th order predictor would normally need K samples from the previous frame in order the predict the current frame's first sample. Since samples from previous frames may not be used, the encoder has either to assume zeros, or to transmit the first K original samples directly, starting the prediction at position K + 1. As a result, compression at the beginning of random access frames would be poor. In order to minimize this problem, the codec uses progressive prediction, which makes use of as many available samples as possible. While it is of course not feasible to predict the first sample of a random access frame, we can use first-order prediction for the second sample, second-order prediction for the third sample, and so forth, until the samples from position K + 1 on are predicted using the full K-th order predictor. Since the predictor
  • Joint Channel Coding Joint channel coding can be used to exploit dependencies between the two channels of a stereo signal, or between any two channels of a multi ⁇
  • each block can be carried out by comparison of the individual signals, depending on which two signals can be coded most efficiently (see Figure 9).
  • Such prediction with switched difference coding is beneficial in cases where two channels are very similar.
  • the channels can be rearranged by the encoder in order to assign suitable channel pairs.
  • Lossless audio codec also supports a more complex scheme for exploiting interchannel redundancy between arbitrary channels of multichannel signals.
  • the residual values e(n) are entropy coded using
  • Rice codes For each block, either all values can be encoded using the same Rice code, or the block can be further divided into four parts, each encoded with a different Rice code.
  • the indices of the applied codes have to be transmitted, as shown in Figure 1. Since there are different ways to determine the optimal Rice code for a given set of data, it is up to the encoder to select suitable codes depending on the statistics of the residual.
  • the encoder can use a more complex and efficient coding scheme called BGMC (Block Gilbert-Moore Codes).
  • BGMC Block Gilbert-Moore Codes
  • the encoding of residuals is accomplished by splitting the distribution in two categories ( Figure 10): Residuals that belong to a central region of the
  • the BGMC encoder splits them into LSB and MSB components first, then it encodes MSBs using block Gilbert-Moore (arithmetic) codes, and finally it transmits LSBs using direct fixed-lengths codes. Both parameters emax and the number of directly transmitted LSBs are selected such that they only slightly affect the coding efficiency of this scheme, while making it significantly less complex. [Compression Results!
  • the lossless audio codec is compared with two of the most popular programs for lossless audio compression:
  • the open-source codec FLAC which uses forward-adaptive prediction as well, and Monkey's Audio (MAC 3.97), a backward-adaptive codec as the current state-of-the-art algorithm in terms of compression.
  • Both codecs were run with options providing maximum compression (flac -8 and mac-c4000).
  • the results for the encoder were determined for a medium compression level (with the prediction order restricted to K _ 60) and a maximum compression level (K _ 1023), both with random access of 500 ms.
  • the tests were conducted on a 1.7 GHz Pentium-M system, with 1024 MB of memory. It comprises nearly 1 GB of stereo waveform data with sampling rates of 48, 96, and 192 kHz, and resolutions of 16 and 24 bits.
  • the compression ratio is defined as
  • the codec is designed to offer a large range of complexity levels. While the maximum level achieves the highest compression at the expense of slowest encoding and decoding speed, the faster medium level only slightly degrades compression, but decoding is significantly less complex than for the maximum level (around 5% CPU load for 48 kHz material). Using a low- complexity level (K _ 15, Rice coding) degrades compression by only 1-1.5% compared to the medium level, but the decoder complexity is further reduced by a factor of three (less than 2% CPU load for 48 kHz material). Thus, audio data can be decoded even on hardware with very low computing power. While the encoder complexity may be increased by both higher maximum orders and a more elaborate block switching algorithm (in accordance with embodiments), the decoder may be affected by a higher average predictor order.
  • the present invention is related the syntax which is comprised in encoded bit stream.
  • the syntax is as bellows;
  • the block_switching field is extended from 1 to 2 bits, the max_order field is extended from 8 to 10 bits.
  • the framejength and user_frame_length fields are merged, resulting in a framejength field of 16 bits, while the user_frame_length field is removed.
  • Frame Data If block switching is used, the bs_info field is added. Depending on the value of block_switching, it has 8, 16, or 32 bits. The first bit of a CPE's bs_info field holds the independent_bs flag. The number of blocks is implicitly derived from bsjnfo as well. If block_switching is off, there is no bs_info field, thus blocks is one and independent_bs is zero.
  • Block Header The short_blocks field is removed, since block switching information is completely transmitted on frame level (bs_info, see previous paragraph).
  • the opt_order field is extended to a maximum of 10 bits (previously 8 bits).

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Abstract

L'invention concerne un procédé et un appareil de codage et de décodage d'un signal audio. Une pluralité de canaux d'une trame de données audio est commutée indépendamment lorsque les canaux ne sont pas corrélés les uns avec les autres ou est commuté de façon synchrone lorsque les canaux sont corrélés les uns avec les autres. Chaque canal est subdivisé en une pluralité de blocs et au moins deux des blocs subdivisés ont des longueurs différentes. Les canaux sont entrelacés lorsqu'ils sont corrélés et ils ne sont pas entrelacés lorsqu'ils ne sont pas corrélés, chaque canal est subdivisé de façon hiérarchique à un ou à plusieurs niveaux de commutation de blocs et chaque bloc a pour résultat une subdivision d'un bloc surordonné de longueur double.
PCT/KR2005/002291 2005-07-11 2005-07-16 Appareil et procede de codage et de decodage de signal audio WO2007011079A1 (fr)

Priority Applications (93)

Application Number Priority Date Filing Date Title
PCT/KR2005/002291 WO2007011079A1 (fr) 2005-07-16 2005-07-16 Appareil et procede de codage et de decodage de signal audio
US11/481,933 US7966190B2 (en) 2005-07-11 2006-07-07 Apparatus and method for processing an audio signal using linear prediction
US11/481,916 US8108219B2 (en) 2005-07-11 2006-07-07 Apparatus and method of encoding and decoding audio signal
US11/481,942 US7830921B2 (en) 2005-07-11 2006-07-07 Apparatus and method of encoding and decoding audio signal
US11/481,926 US7949014B2 (en) 2005-07-11 2006-07-07 Apparatus and method of encoding and decoding audio signal
US11/481,917 US7991272B2 (en) 2005-07-11 2006-07-07 Apparatus and method of processing an audio signal
US11/481,929 US7991012B2 (en) 2005-07-11 2006-07-07 Apparatus and method of encoding and decoding audio signal
US11/481,930 US8032368B2 (en) 2005-07-11 2006-07-07 Apparatus and method of encoding and decoding audio signals using hierarchical block swithcing and linear prediction coding
US11/481,941 US8050915B2 (en) 2005-07-11 2006-07-07 Apparatus and method of encoding and decoding audio signals using hierarchical block switching and linear prediction coding
US11/481,931 US7411528B2 (en) 2005-07-11 2006-07-07 Apparatus and method of processing an audio signal
US11/481,939 US8121836B2 (en) 2005-07-11 2006-07-07 Apparatus and method of processing an audio signal
US11/481,940 US8180631B2 (en) 2005-07-11 2006-07-07 Apparatus and method of processing an audio signal, utilizing a unique offset associated with each coded-coefficient
US11/481,915 US7996216B2 (en) 2005-07-11 2006-07-07 Apparatus and method of encoding and decoding audio signal
US11/481,927 US7835917B2 (en) 2005-07-11 2006-07-07 Apparatus and method of processing an audio signal
US11/481,932 US8032240B2 (en) 2005-07-11 2006-07-07 Apparatus and method of processing an audio signal
JP2008521306A JP2009500682A (ja) 2005-07-11 2006-07-10 オーディオ信号のエンコーディング及びデコーディング装置及び方法
JP2008521310A JP2009500686A (ja) 2005-07-11 2006-07-10 オーディオ信号のエンコーディング及びデコーディング装置及び方法
JP2008521319A JP2009500693A (ja) 2005-07-11 2006-07-10 オーディオ信号のエンコーディング及びデコーディング装置及び方法
PCT/KR2006/002690 WO2007008012A2 (fr) 2005-07-11 2006-07-10 Dispositif et procédé de traitement d'un signal audio
CN2006800294070A CN101243496B (zh) 2005-07-11 2006-07-10 处理音频信号的装置和方法
PCT/KR2006/002689 WO2007008011A2 (fr) 2005-07-11 2006-07-10 Dispositif et procede de traitement d'un signal audio
PCT/KR2006/002691 WO2007008013A2 (fr) 2005-07-11 2006-07-10 Dispositif et procede destines au codage et au decodage d'un signal audio
CNA2006800289829A CN101238510A (zh) 2005-07-11 2006-07-10 处理音频信号的装置和方法
EP06769219A EP1913584A4 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et de decodage de signal audio
PCT/KR2006/002678 WO2007008000A2 (fr) 2005-07-11 2006-07-10 Appareil et procede d'encodage et de decodage de signal audio
JP2008521308A JP2009500684A (ja) 2005-07-11 2006-07-10 オーディオ信号を処理する方法、オーディオ信号のエンコーディング及びデコーディング装置及び方法
EP06769224A EP1913794A4 (fr) 2005-07-11 2006-07-10 Appareil et procede de traitement d'un signal audio
CNA2006800294174A CN101243489A (zh) 2005-07-11 2006-07-10 编码和解码音频信号的装置和方法
JP2008521305A JP2009500681A (ja) 2005-07-11 2006-07-10 オーディオ信号のエンコーディング及びデコーディング装置及び方法
EP06769218A EP1913589A4 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et de decodage de signal audio
CNA2006800304693A CN101243492A (zh) 2005-07-11 2006-07-10 编码和解码音频信号的装置和方法
EP06769220A EP1913585A4 (fr) 2005-07-11 2006-07-10 Appareil et procede pour traitement du signal audio
CNA2006800305412A CN101243497A (zh) 2005-07-11 2006-07-10 编码和解码音频信号的装置和方法
EP06769223A EP1913587A4 (fr) 2005-07-11 2006-07-10 Dispositif et procede de traitement d'un signal audio
CNA2006800251376A CN101218631A (zh) 2005-07-11 2006-07-10 处理音频信号的装置和方法
PCT/KR2006/002677 WO2007007999A2 (fr) 2005-07-11 2006-07-10 Appareil et procede d'encodage et de decodage de signal audio
JP2008521307A JP2009500683A (ja) 2005-07-11 2006-07-10 オーディオ信号のエンコーディング及びデコーディング装置及び方法
EP06769226A EP1913588A4 (fr) 2005-07-11 2006-07-10 Dispositif et procédé de traitement d'un signal audio
EP06757764A EP1913579A4 (fr) 2005-07-11 2006-07-10 Appareil et procede d'encodage et de decodage de signal audio
PCT/KR2006/002682 WO2007008004A2 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et decodage de signal audio
PCT/KR2006/002687 WO2007008009A1 (fr) 2005-07-11 2006-07-10 Dispositif et procede de traitement d'un signal audio
CNA2006800251395A CN101218629A (zh) 2005-07-11 2006-07-10 处理音频信号的装置和方法
JP2008521318A JP2009500692A (ja) 2005-07-11 2006-07-10 オーディオ信号の処理装置及び方法
EP06769225A EP1911021A4 (fr) 2005-07-11 2006-07-10 Dispositif et procede de traitement d'un signal audio
PCT/KR2006/002680 WO2007008002A2 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et de decodage de signal audio
JP2008521314A JP2009500689A (ja) 2005-07-11 2006-07-10 オーディオ信号の処理装置及び方法
PCT/KR2006/002679 WO2007008001A2 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et de decodage de signal audio
PCT/KR2006/002686 WO2007008008A2 (fr) 2005-07-11 2006-07-10 Appareil et procede de traitement d'un signal audio
EP06757765A EP1913580A4 (fr) 2005-07-11 2006-07-10 Appareil et procede d'encodage et de decodage de signal audio
EP06769227A EP1911020A4 (fr) 2005-07-11 2006-07-10 Dispositif et procede destines au codage et au decodage d'un signal audio
CNA2006800305111A CN101243494A (zh) 2005-07-11 2006-07-10 编码和解码音频信号的装置和方法
CNA200680024866XA CN101218852A (zh) 2005-07-11 2006-07-10 处理音频信号的装置和方法
PCT/KR2006/002681 WO2007008003A2 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et de decodage de signal audio
EP06757767A EP1913582A4 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et decodage de signal audio
EP06757768A EP1913583A4 (fr) 2005-07-11 2006-07-10 Appareil et procede pour traitement du signal audio
JP2008521313A JP2009500688A (ja) 2005-07-11 2006-07-10 オーディオ信号の処理装置及び方法
CNA2006800304797A CN101243493A (zh) 2005-07-11 2006-07-10 编码和解码音频信号的装置和方法
JP2008521309A JP2009500685A (ja) 2005-07-11 2006-07-10 オーディオ信号のエンコーディング及びデコーディング装置及び方法
PCT/KR2006/002685 WO2007008007A1 (fr) 2005-07-11 2006-07-10 Appareil et procede pour traitement du signal audio
JP2008521315A JP2009500690A (ja) 2005-07-11 2006-07-10 オーディオ信号の処理装置及び方法
CN2006800251380A CN101218628B (zh) 2005-07-11 2006-07-10 编码和解码音频信号的装置和方法
EP06769222A EP1908058A4 (fr) 2005-07-11 2006-07-10 Appareil et procede de traitement d'un signal audio
JP2008521311A JP2009500687A (ja) 2005-07-11 2006-07-10 オーディオ信号の処理装置及び方法
CNA2006800305499A CN101243495A (zh) 2005-07-11 2006-07-10 编码和解码音频信号的装置和方法
PCT/KR2006/002683 WO2007008005A1 (fr) 2005-07-11 2006-07-10 Appareil et procede pour traitement du signal audio
CNA200680028892XA CN101238509A (zh) 2005-07-11 2006-07-10 处理音频信号的装置和方法
PCT/KR2006/002688 WO2007008010A1 (fr) 2005-07-11 2006-07-10 Appareil et procede de traitement d'un signal audio
EP06757766A EP1913581A4 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et de decodage de signal audio
CN2006800252699A CN101218630B (zh) 2005-07-11 2006-07-10 处理音频信号的装置和方法
JP2008521317A JP2009500691A (ja) 2005-07-11 2006-07-10 オーディオ信号の処理装置及び方法
JP2008521316A JP2009510810A (ja) 2005-07-11 2006-07-10 オーディオ信号の処理装置及び方法
US12/232,527 US7962332B2 (en) 2005-07-11 2008-09-18 Apparatus and method of encoding and decoding audio signal
US12/232,526 US8010372B2 (en) 2005-07-11 2008-09-18 Apparatus and method of encoding and decoding audio signal
US12/232,590 US8055507B2 (en) 2005-07-11 2008-09-19 Apparatus and method for processing an audio signal using linear prediction
US12/232,593 US8326132B2 (en) 2005-07-11 2008-09-19 Apparatus and method of encoding and decoding audio signal
US12/232,591 US8255227B2 (en) 2005-07-11 2008-09-19 Scalable encoding and decoding of multichannel audio with up to five levels in subdivision hierarchy
US12/232,595 US8417100B2 (en) 2005-07-11 2008-09-19 Apparatus and method of encoding and decoding audio signal
US12/232,662 US8510120B2 (en) 2005-07-11 2008-09-22 Apparatus and method of processing an audio signal, utilizing unique offsets associated with coded-coefficients
US12/232,659 US8554568B2 (en) 2005-07-11 2008-09-22 Apparatus and method of processing an audio signal, utilizing unique offsets associated with each coded-coefficients
US12/232,658 US8510119B2 (en) 2005-07-11 2008-09-22 Apparatus and method of processing an audio signal, utilizing unique offsets associated with coded-coefficients
US12/232,744 US8032386B2 (en) 2005-07-11 2008-09-23 Apparatus and method of processing an audio signal
US12/232,739 US8155152B2 (en) 2005-07-11 2008-09-23 Apparatus and method of encoding and decoding audio signal
US12/232,741 US8149877B2 (en) 2005-07-11 2008-09-23 Apparatus and method of encoding and decoding audio signal
US12/232,740 US8149876B2 (en) 2005-07-11 2008-09-23 Apparatus and method of encoding and decoding audio signal
US12/232,748 US8155153B2 (en) 2005-07-11 2008-09-23 Apparatus and method of encoding and decoding audio signal
US12/232,747 US8149878B2 (en) 2005-07-11 2008-09-23 Apparatus and method of encoding and decoding audio signal
US12/232,743 US7987008B2 (en) 2005-07-11 2008-09-23 Apparatus and method of processing an audio signal
US12/232,734 US8155144B2 (en) 2005-07-11 2008-09-23 Apparatus and method of encoding and decoding audio signal
US12/232,781 US7930177B2 (en) 2005-07-11 2008-09-24 Apparatus and method of encoding and decoding audio signals using hierarchical block switching and linear prediction coding
US12/232,782 US8046092B2 (en) 2005-07-11 2008-09-24 Apparatus and method of encoding and decoding audio signal
US12/232,784 US7987009B2 (en) 2005-07-11 2008-09-24 Apparatus and method of encoding and decoding audio signals
US12/232,783 US8275476B2 (en) 2005-07-11 2008-09-24 Apparatus and method of encoding and decoding audio signals
US12/314,891 US8065158B2 (en) 2005-07-11 2008-12-18 Apparatus and method of processing an audio signal

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Non-Patent Citations (3)

* Cited by examiner, † Cited by third party
Title
DAI YANG ET AL.: "A lossless audio compression scheme with random access property", ICASSP 2004, vol. 3, 17 May 2004 (2004-05-17) - 21 May 2004 (2004-05-21), pages 1016 - 1019, XP010718365 *
LIEBCHEN T.: "An introduction to MPEG-4 audio lossless coding", 2004 INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING (ICASSP '04), vol. 3, 17 May 2004 (2004-05-17) - 21 May 2004 (2004-05-21), pages 1012 - 1015, XP010718364 *
MORIYA T. ET AL.: "Extended linear prediction tools for lossless audio coding", ICASSP 2004, vol. 3, 17 May 2004 (2004-05-17) - 21 May 2004 (2004-05-21), pages 1008 - 1011, XP010718363 *

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