Patent Grant
7725310
The present invention relates to encoding and decoding of broadband signals, in particular audio signals. The invention relates both to the encoder and the decoder, and to an audio stream encoded according to the invention and a data storage medium on which such an audio stream has been stored.
When transmitting broadband signals, e.g. audio signals such as speech, compression or encoding techniques are used to reduce the bandwidth or bit rate of the signal.
In the sinusoidal analyser 130 of
In such sinusoidal encoding, it is usual to transmit frequency information for the tracks formed in the encoder. This can be done in a simple manner and with relatively low costs, because tracks only have a slowly varying frequency. Frequency information can therefore be transmitted efficiently by time-differential encoding. In general, amplitude can also be encoded differentially over time.
In contrast to frequency, phase changes more rapidly with time. If the frequency is (substantially) constant, the phase will change (substantially) linearly with time, and frequency changes will result in corresponding phase deviations from the linear course. As a function of the track segment index, phase will have an approximately linear behavior. Transmission of encoded phase is therefore more complicated. However, when transmitted, phase is limited to the range {−π;π}, i.e. the phase is “wrapped”, as provided by the Fourier transform. Because of this modulo 2π representation of phase, the structural inter-frame relation of the phase is lost and, at first sight, appears to be a random variable.
However, since the phase is the integral of the frequency, the phase is redundant and, in principle, does not need to be transmitted. This reduces the bit rate significantly. In the decoder, the phase is recovered by a process which is called phase continuation.
In phase continuation, only the encoded frequency is transmitted, and the phase is recovered at the decoder from the frequency data by exploiting the integral relation between phase and frequency. It is known, however, that when phase continuation is used, the phase cannot be perfectly recovered. If frequency errors occur, e.g. due to measurement errors in the frequency or due to quantization noise, the phase, which is being reconstructed by using the integral relation, will typically show an error having the character of drift. This is because frequency errors have an approximately random character. Low-frequency errors are amplified by integration, and consequently the recovered phase will tend to drift away from the actually measured phase. This leads to audible artifacts.
This is illustrated in
Thus, it can be seen that in phase continuation, the recovered phase is a low-frequency signal itself because the recovered phase is the integral of a low-frequency signal. However, the noise introduced in the reconstruction process is also dominant in this low-frequency range. It is therefore difficult to separate these sources with a view to filtering the noise n introduced during encoding.
Furthermore, in phase continuation, only the first sinusoid of each track is transmitted for each track in order to save bit rate. Each subsequent phase is calculated from the initial phase and frequencies of the track. Since the frequencies are quantized and not always estimated very accurately, the continuous phase will deviate from the measured phase. Experiments show that phase continuation degrades the quality of an audio signal.
European Patent Application 02080002.5 (PHNL021216) addresses these problems by proposing a joint frequency/phase quantizer, where the measured phases of a sinusoidal track, which have values between −π and π, are unwrapped by using the measured frequencies and linking information, resulting in monotonic increasing unwrapped phases along a track. In the encoder, the unwrapped phases are quantized by using an Adaptive Differential Pulse Code Modulation (ADPCM) quantizer and transmitted to the decoder. The decoder derives the frequencies and the phases of a sinusoidal track from the unwrapped phase trajectory.
As an example, the ADPCM quantizer can be configured as described below. For the first continuation of a track, the unwrapped phase is quantized in accordance with Table 1.
The quantization boundaries are defined in accordance with this table by: {−∞; 2·T (r=1), 0, 2·T (r=2), ∞}. For each consecutive continuation, the tables are scaled. If the representation level is in the outer level, the tables are multiplied by 21/2, making the quantization accuracy coarser. Otherwise, the representation levels are in the inner level and the tables are scaled by 2−1/4, making the quantization accuracy finer. Furthermore, there is an upper and lower boundary to the inner level, namely 3π/4 and π/64.
The quantization of the unwrapped phase trajectory is a continuous process in the above methods, where the quantization accuracy is adapted along the track. Therefore, in order to decode a track, the decoding process has to start from the birth or starting point of a track, i.e. the decoder can only de-quantize a complete track and it is not possible to decode a part of the track. Therefore, special methods enabling random-access have to be added to the encoder and decoder. Random-access may e.g. be used to ‘skip’ or ‘fast forward’ in an audio signal.
A first straightforward way of performing random access is to define random-access frames (or refresh points) in the encoder/quantizer and re-start the ADPCM quantizer in the decoder at these random-access frames. For the random-access frame, the initial tables are used. Therefore, refreshes are as expensive in bits as normal births. However, a drawback of this approach is that the quantization tables and thus the quantization accuracy have to be adapted again from the random-access frame and onwards. Therefore, initially, the quantization accuracy might be too coarse, resulting in a discontinuity in the track, or too fine, resulting in large quantization errors. This leads to a degradation of the audio quality compared to the decoded signals without the use of random-access frames.
A second straightforward way is to transmit all states of the ADPCM quantizer (that is the quantization accuracy and the memories in the predictor as mentioned in European Patent Application 02080002.5 (PHNL021216). The quantizer will then have similar output with or without random-access frames. In this way, the sound quality will hardly suffer. However, the additional bit rate to transmit all this information will be considerable. Especially since the contents of the memories of the predictor have to be quantized according to the quantization accuracy of the ADPCM quantizer.
The present invention addresses these problems.
The present invention provides a method of encoding a broadband signal, in particular an audio signal or a speech signal, using a low bit rate. More specifically, the invention provides a method of encoding an audio signal, the method comprising the steps of: providing a respective set of sampled signal values for each of a plurality of sequential time segments; analyzing the sampled signal values to determine one or more sinusoidal components for each of the plurality of sequential segments; linking sinusoidal components across a plurality of sequential segments to provide sinusoidal tracks, each track comprising a number of frames; and generating an encoded signal including sinusoidal codes comprising a representation level for zero or more frames and where some of these codes comprise a phase, a frequency and a quantization table for a given frame when the given frame is designated as a random-access frame.
In this way, random-access is enabled, e.g. allowing skipping through a track, etc., while avoiding the long adaptation of the quantization accuracy in a quantizer, e.g. an ADPCM quantizer, of the prior art, as (some) of the quantization state is transmitted (in the form of the quantization table) to the encoder.
Furthermore, the quantization table is adapted to be faster as compared with the first straightforward method that uses the default initial table. Additionally, as compared with the second straightforward method, the present invention results in a lower bit rate.
The present invention offers a good compromise between the two (straightforward) methods, by transmitting only the quantization accuracy, thereby providing a good quality at a low bit rate.
In a preferred embodiment, each quantization table is represented by an index where the index is transmitted from the encoder to the decoder at a random-access frame instead of the quantization table. The index may e.g. be generated or represented by using Huffman coding.
Preferably, the phase (φ) and the frequency (ω) for a random-access frame are the measured phase and the measured frequency in the refresh frame quantized according to the default method used for quantising a starting point of a track. These phases and frequencies will also be referred to as φ(0) and ω(0), respectively.
a illustrates the relationship between phase and frequency in prior-art systems;
b illustrates the relationship between phase and frequency in audio systems using phase encoding;
a and 3b show a preferred embodiment of a sinusoidal encoder component of the audio encoder of
a and 5b show a preferred embodiment of a sinusoidal synthesizer component of the audio player of
a and 7b illustrate the information sent from the encoder and received at the decoder according to the prior art and to the present invention, respectively.
Preferred embodiments of the invention will now be described with reference to the accompanying drawings wherein like components have been accorded like reference numerals and, unless otherwise stated, perform like functions.
In both the prior art and the preferred embodiment of the present invention, the audio encoder 1 samples an input audio signal at a certain sampling frequency, resulting in a digital representation x(t) of the audio signal. The encoder 1 then separates the sampled input signal into three components: transient signal components, sustained deterministic components, and sustained stochastic components. The audio encoder 1 comprises a transient encoder 11, a sinusoidal encoder 13 and a noise encoder (NA) 14.
The transient encoder 11 comprises a transient detector (TD) 110, a transient analyzer (TA) 111 and a transient synthesizer (TS) 112. First, the signal x(t) enters the transient detector 110. This detector 110 estimates if there is a transient signal component and its position. This information is fed to the transient analyzer (TA) 111. If the position of a transient signal component is determined, the transient analyzer (TA) 111 tries to extract (the main part of) the transient signal component It matches a shape function to a signal segment preferably starting at an estimated start position, and determines content underneath the shape function, by employing, for example, a (small) number of sinusoidal components. This information is contained in the transient code CT, and more detailed information on generating the transient code CT is provided in WO 01/69593.
The transient code CT is furnished to the transient synthesizer (TS) 112. The synthesized transient signal component is subtracted from the input signal x(t) in subtractor 16, resulting in a signal x1. A gain control mechanism GC (12) is used to produce x2 from x1.
The signal x2 is furnished to the sinusoidal encoder 13 where it is analyzed in a sinusoidal analyzer (SA) 130, which determines the (deterministic) sinusoidal components. It will therefore be seen that, while the presence of the transient analyzer is desirable, it is not necessary and the invention can be implemented without such an analyzer. Alternatively, as mentioned above, the invention can also be implemented with, for example, a harmonic complex analyzer. In brief, the sinusoidal encoder encodes the input signal x2 as tracks of sinusoidal components linked from one frame segment to the next.
Referring now to
The sinusoidal codes CS ultimately produced by the analyzer 130 include phase information, and frequency is reconstructed from this information in the decoder, as is mentioned in European Patent Application 02080002.5 (PHNL021216). According to the present invention, a quantization table (Q) or preferably an index (IND) representing the quantization table (Q) is produced by the analyzer 130 instead of a representation level r when the given sub-frame being processed is a random-access frame, as will be explained in greater detail with reference to
As mentioned above, however, the measured phase φ(k) is wrapped, which means that it is restricted to a modulo 2π representation. Therefore, in the preferred embodiment, the analyzer comprises a phase unwrapper (PU) 44 where the modulo 2π phase representation is unwrapped to expose the structural inter-frame phase behavior ψ for a track. As the frequency in sinusoidal tracks is nearly constant it will be seen that the unwrapped phase ψ will typically be a nearly linearly increasing (or decreasing) function and this makes cheap transmission of phase, i.e. with low bit rate, possible. The unwrapped phase ψ is provided as input to a phase encoder (PE) 46, which provides, as output, quantized representation levels r suitable for being transmitted (when a given sub-frame is not a random-access frame).
Referring now to the operation of the phase unwrapper 44, as mentioned above, instantaneous phase ψ and instantaneous frequency Ω for a track are related by:
where T0 is a reference time instant.
A sinusoidal track in frames k=K, K+1 . . . K+L−1 has measured frequencies ω(k) (expressed in radians per second) and measured phases φ(k) (expressed in radians). The distance between the centres of the frames is given by U (update rate expressed in seconds). The measured frequencies are supposed to be samples of the assumed underlying continuous-time frequency track Ω with ω(k)=Ω(kU) and, similarly, the measured phases are samples of the associated continuous-time phase track ψ with φ(k)=ψ(kU) mod (2π). For sinusoidal encoding, it is assumed that Ω is a nearly constant function.
Assuming that the frequencies are nearly constant within a segment, Equation 1 can be approximated as follows:
It will therefore be seen that, knowing the phase and frequency for a given segment and the frequency of the next segment, it is possible to estimate an unwrapped phase value for the next segment, and so on for each segment in a track.
In the preferred embodiment, the phase unwrapper determines an unwrap factor m(k) at time instant k:
ψ(kU)=φ(k)+m(k)2π (3)
The unwrap factor m(k) tells the phase unwrapper 44 the number of cycles which has to be added to obtain the unwrapped phase.
Combining equations 2 and 3, the phase unwrapper determines an incremental unwrap factor e(k) as follows:
2πe(k)=2π{m(k)−m(k−1)}={ω(k)+ω(k−1)}U/2−{φ(k)−φ(k−1)}
where e should be an integer. However, due to measurement and model errors, the incremental unwrap factor will not be an integer exactly, so:
e(k)=round([{ω(k)+ω(k−1)}U/2−{φ(k)−φ(k−1)}]/(2π))
assuming that the model and measurement errors are small.
Having the incremental unwrap factor e, the m(k) from equation (3) is calculated as the cumulative sum where, without loss of generality, the phase unwrapper starts in the first frame K with m(K)=0, and from m(k) and φ(k), the (unwrapped) phase ψ(kU) is determined.
In practice, the sampled data ψ(kU) and Ω(kU) are distorted by measurement errors:
φ(k)=ψ(kU)+ε1(k),
ω(k)=Ω(kU)+ε2(k),
where ε1 and ε2 are the phase and frequency errors, respectively. In order to prevent the determination of the unwrap factor becoming ambiguous, the measurement data needs to be determined with sufficient accuracy. Thus, in the preferred embodiment, tracking is restricted so that:
δ(k)=e(k)−[{ω(k)+ω(k−1)}U/2−{φ(k)−φ(k−1)}]/(2π)<δ0
where δ is the error in the rounding operation. The error δ is mainly determined by the errors in ω due to the multiplication with U. Assume that ω is determined from the maxima of the absolute value of the Fourier transform from a sampled version of the input signal with sampling frequency Fs and that the resolution of the Fourier transform is 2π/La with La being the analysis size. In order to be within the considered bound, we have:
This means that the analysis size should be few times larger than the update size in order for unwrapping to be accurate, e.g., setting δ0=¼, the analysis size should be four times the update size (neglecting the errors ε1 in the phase measurement).
The second precaution, which can be taken to avoid decision errors in the round operation, is to define tracks appropriately. In the tracking unit 42, sinusoidal tracks are typically defined by considering amplitude and frequency differences. Additionally, it is also possible to account for phase information in the linking criterion. For instance, we can define the phase prediction error e as the difference between the measured value and the predicted value {tilde over (φ)} according to
ε={φ(k)−{tilde over (φ)}(k)}mod2π
where the predicted value can be taken as
{tilde over (φ)}(k)=φ(k−1)+{ω(k)−ω(k−1)}U/2
Thus, preferably the tracking unit (TRA) 42 forbids tracks where ε is larger than a certain value (e.g. ε>π/2), resulting in an unambiguous definition of e(k).
Additionally, the encoder may calculate the phases and frequencies such as will be available in the decoder. If the phases or frequencies which will become available in the decoder differ too much from the phases and/or frequencies such as are present in the encoder, it may be decided to interrupt a track, i.e. to signal the end of a track and start a new one using the current frequency and phase and their linked sinusoidal data.
The sampled unwrapped phase ψ(kU) produced by the phase unwrapper (PU) 44 is provided as input to phase encoder (PE) 46 to produce the set of representation levels r (or according to the present invention, a quantization table (Q) or an index (IND) representing the quantization table (Q) when the given sub-frame being processed/transmitted is a random-access frame. Techniques for efficient transmission of a generally monotonically changing characteristic such as the unwrapped phase are known.
b illustrates a preferred embodiment of the phase encoder (PE) 46. In this preferred embodiment, Adaptive Differential Pulse Code Modulation (ADPCM) is employed. Here, a predictor (PF) 48 is used to estimate the phase of the next track segment and encode the difference only in a quantizer (QT) 50. Since ψ is expected to be a nearly linear function and, also for reasons of simplicity, the predictor 48 is chosen as a second-order filter of the form:
y(k+1)=2x(k)−x(k−1)
where x is the input and y is the output. It will be seen, however, that it is also possible to take other functional relations (including higher-order relations) and to include (backward or forward) adaptation of the filter coefficients. In the preferred embodiment, a backward adaptive control mechanism (QC) 52 is used for simplicity to control the quantizer (QT) 50. Forward adaptive control is possible as well but would require extra bit rate.
As will be seen, initialization of the encoder (and decoder) for a track starts with knowledge of the start phase φ(0) and frequency ω(0). These are quantized and transmitted by a separate mechanism. Additionally, the initial quantization step used in the quantization controller (QC) 52 of the encoder and the corresponding controller 62 in the decoder,
The start frequency of the unwrapped phase is known, both in the encoder and in the decoder. The quantization accuracy is chosen on the basis of this frequency. For the unwrapped phase trajectories beginning with a low frequency, a more accurate quantization grid, i.e. a higher resolution, is chosen than for an unwrapped phase trajectory beginning with a higher frequency.
In the ADPCM quantizer, the unwrapped phase ψ(k), where k represents the number in the track, is predicted/estimated from the preceding phases in the track. The difference between the predicted phase {tilde over (ψ)}(k) and the unwrapped phase ψ(k) is then quantized and transmitted. The quantizer is adapted for every unwrapped phase in the track. When the prediction error is small, the quantizer limits the range of possible values and the quantization can become more accurate. On the other hand, when the prediction error is large, the quantizer uses a coarser quantization.
The quantizer Q in
Δ(k)=ψ(k)−{tilde over (ψ)}(k)
The prediction error A can be quantized by using a look-up table. For this purpose, a table Q is maintained. For example, for a 2-bit ADPCM quantizer, the initial table for Q may look like the table shown in Table 2.
The quantization is done as follows. The prediction error Δ is compared with the boundaries b, such that the following equation is satisfied:
b1i<Δ≦bui
From the value of i, which satisfies the above relation, the representation level r is computed by r=i.
The associated representation levels are stored in representation table R, which is shown in Table 3.
The entries of tables Q and R are multiplied by a factor c for the quantization of the next sinusoidal component in the track.
{tilde under (Q)}(k+1)={tilde under (Q)}(k)·c
R(k+1)=R(k)·c
During the decoding of a track, both tables are scaled in accordance with the generated representation levels r. If r is either 1 or 2 (inner level) for the current sub-frame, then the scale factor c for the quantization table is set to
c=2−1/4
Since c<1, the frequency and phase of the next sinusoid in a track become more accurate. If r is 0 or 3 (outer level), the scale factor is set to
c=21/2
Since c>1, the quantization accuracy for the next sinusoid in a track decreases. Using these factors, one up-scaling can be made undone by two down-scalings. The difference in upscale and downscale factors results in a fast onset of an up-scaling, whereas a corresponding downscaling requires two steps.
In order to avoid very small or very large entries in the quantization table, the adaptation is only done if the absolute value of the inner level is between π/64 and 3π/4. In case the inner level is less than or equal to π/64 or greater than or equal to 3π/4 the scale factor c is set to 1.
In the decoder, only table R has to be maintained to convert the received representation levels r to a quantized prediction error. This de-quantization operation is performed by block (DQ) 60 in
Using the above settings, the quality of the reconstructed sound needs improvement. Different initial tables for unwrapped phase tracks, depending on the start frequency, may be used. This yields a better sound quality. This is done as follows. The initial tables Q and R are scaled on the basis of a first frequency of the track. In Table 4, the scale factors are given together with the frequency ranges. If the first frequency of a track lies in a certain frequency range, the appropriate scale factor is selected, and the tables R and Q are divided by that scale factor. The end-points may also depend on the first frequency of the track. In the decoder, a corresponding procedure is performed in order to start with the correct initial table R.
Table 4 shows an example of frequency-dependent scale factors and corresponding initial tables Q and R for a 2-bit ADPCM quantizer. The audio frequency range 0-22050 Hz is divided into four frequency sub-ranges. It can be seen that the phase accuracy is improved in the lower frequency ranges relative to the higher frequency ranges.
The number of frequency sub-ranges and the frequency-dependent scale factors may vary and can be chosen to fit the individual purpose and requirements. As described above, the frequency-dependent initial tables Q and R in table 4 may be upscaled and down-scaled dynamically to adapt to the evolution in phase from one time segment to the next.
In e.g. a 3-bit ADPCM quantizer, the initial boundaries of the eight quantization intervals defined by the 3 bits can be defined as follows:
Q={−∞−1.41 −0.707 −0.35 0 0.35 0.707 1.41 ∞}, and can have minimum grid size π/64, and a maximum grid size π/2. The representation table R may look like:
R={−2.117, −1.0585, −0.5285, −0.1750, 0.1750, 0.5285, 1.0585, 2.117}. A similar frequency-dependent initialization of the table Q and R as shown in Table 4 may be used in this case.
So far, the process has been described in the same way as in European Patent Application 02080002.5 (PHNL021216).
According to the present invention, quantizer (QT) 50, predictor (PF) 48 and backward adaptive control mechanism (QC) 52 may further receive a (external) trigger signal (Trig.) indicating that the given frame being processed is a random-access frame. When no trigger signal (Trig.) is received, the process functions normally and only representation levels r are transmitted to the decoder. When a trigger (Trig.) is received (signifying a random-access frame), no representation levels r are transmitted but, instead, the quantization table (Q) or an index (IND) representing the quantization table (Q) is transmitted, together with the current phase (φ(0)) and the current frequency (ω0)).
By proper setting of the quantizer parameters, only a limited number of quantization tables are possible. For the example given in Table 1, there are only 22 possible quantization tables, which are listed below in Table 5 together with an index number. The entries in Table 5 are rounded values of
where k ranges from −23, −22, . . . , 5, 6.
Consequently, in a preferred embodiment, in order to reduce the amount of data transmitted, only an index representing/identifying/indicating the given quantization table (Q) is transmitted to the encoder where the index is used to retrieve the appropriate quantization table used as the initial table, which is explained in greater detail with reference to
Preferably, an index is generated by using the well-known Huffman coding. For table 5, such a Huffman coding-based index may be as listed in table 6 below:
In a preferred embodiment, instead of sending a given quantization table or quantization state (e.g. 19:T1=−0.1577; T2=−0.0394; T3=0.0394; T4=0.1577), only the index (IND) (e.g. 010001) is transmitted, thereby saving bit rate. This index is then used at the decoder to retrieve the proper quantization table (e.g. 19), which is then used according to the present invention.
In this way, random-access is enabled while avoiding long adaptation for high accuracy in the quantizer, because no re-starting of the quantizer is needed as the current accuracy of the quantization table is stored and transmitted to the decoder (either directly, by transmitting the given quantization table (Q), or indirectly, by transmitting an index (IND) referencing/uniquely identifying/indicating the given quantization table (Q). Furthermore, the quantization table is adapted to be faster and/or a lower bit rate is obtained.
Random-access frames may e.g. be selected or identified by selecting every N'th frame during a track, using audio analysis to select appropriate points, etc. For each random-access frame, the trigger signal is provided to the quantizer (QT) 50 (and (PF) 48 and (QC) 52) when a random-access frame is being processed.
From the sinusoidal code Cs generated with the sinusoidal encoder, the sinusoidal signal component is reconstructed by a sinusoidal synthesizer (SS) 131 in the same manner as will be described for the sinusoidal synthesizer (SS) 32 of the decoder. This signal is subtracted in subtractor 17 from the input x2 to the sinusoidal encoder 13, resulting in a residual signal x3. The residual signal x3 produced by the sinusoidal encoder 13 is passed to the noise analyzer 14 of the preferred embodiment which produces a noise code CN representative of this noise, as described in, for example, international patent application No. PCT/EP00/04599.
Finally, in a multiplexer 15, an audio stream AS is constituted which includes the codes CT, CS and CN. The audio stream AS is furnished to e.g. a data bus, an antenna system, a storage medium, etc.
The sinusoidal code CS including the information encoded by the analyzer 130 is used by the sinusoidal synthesizer 32 to generate signal yS. Referring now to
As illustrated in
In the preferred embodiment, a filtering unit (FR) 58 approximates the differentiation, which is necessary to obtain the frequency {circumflex over (ω)} from the unwrapped phase by procedures as forward, backward or central differences. This enables the decoder to produce as output the phases {circumflex over (ψ)} and frequencies {circumflex over (ω)} usable in a conventional manner to synthesize the sinusoidal component of the encoded signal.
At the same time, as the sinusoidal components of the signal are being synthesized, the noise code CN is fed to a noise synthesizer NS 33, which is mainly a filter, having a frequency response approximating the spectrum of the noise. The NS 33 generates reconstructed noise yN by filtering a white noise signal with the noise code CN. The total signal y(t) comprises the sum of the transient signal yT and the product of any amplitude decompression (g) and the sum of the sinusoidal signal yS and the noise signal yN. The audio player comprises two adders 36 and 37 to sum respective signals. The total signal is furnished to an output unit 35, which is e.g. a speaker.
According to the present invention, for random-access frames, the transmitted quantization table (Q) or an index (IND) is received from the encoder instead of the representation levels r. The indication that the received frame is a random-access frame may e.g. be implemented by adding an additional field in the bit stream syntax comprising the appropriate index e.g. as shown in Table 6, thereby identifying the specific quantization table (Q) to be used. The index is obtained from the Huffman code. This index indicates the table that is used for the ADPCM, as shown in Table 5. This table includes all possible quantization tables Q. The number depends on the up-scale and down-scale factors and the minimum and maximum values of the inner level. If the current frame is a random-access frame, meaning that sub-frame K includes, for each sinusoid in the sub-frame, the additional field of the bit stream syntax having a value of a Huffman code (supplied to (QC) 62, (DQ) 60 and (PF) 64 as the trigger signal (Trig.)). Furthermore, sub-frame K also includes the directly quantized amplitude, frequency and phase for each sinusoid as specified by the encoder. The field of the bit stream syntax is Huffman decoded and the appropriate table T is selected in accordance with Table 5. This table is then used for the de-quantizer (DQ) (60) in the next sub-frame (K+1). The prediction filter (PF) 64 is re-initialized for sub-frame K+1 in the same way as is done for the first continuation:
ψr(K−1)=φ(K)−ω(K)·U,
where U is the update interval. Here φ is the phase and ω is the frequency transmitted in the sub-frame K. The decoding continues in the traditional fashion as described above.
a and 7b illustrate the information sent from the encoder and received at the decoder according to the prior art and to the present invention, respectively.
b illustrates a number of frames (701, 702, 703) shown with their frame number and frequency according to the present invention, as well as the information or parameters that are transmitted from an encoder to a decoder for each (sub-)frame. As can be seen, the initial phase (φ(0)) and initial frequency (ω(0)) are transmitted for the birth or start of track frame (701), similarly as in
| Number | Date | Country | Kind |
|---|---|---|---|
| 03103774 | Oct 2003 | EP | regional |
| Filing Document | Filing Date | Country | Kind | 371c Date |
|---|---|---|---|---|
| PCT/IB2004/051963 | 10/4/2004 | WO | 00 | 4/10/2006 |
| Publishing Document | Publishing Date | Country | Kind |
|---|---|---|---|
| WO2005/036529 | 4/21/2005 | WO | A |
| Number | Name | Date | Kind |
|---|---|---|---|
| 4937873 | McAulay et al. | Jun 1990 | A |
| 6292777 | Inoue et al. | Sep 2001 | B1 |
| 20020007268 | Oomen et al. | Jan 2002 | A1 |
| 20030135374 | Hardwick | Jul 2003 | A1 |
| 20040039568 | Toyama et al. | Feb 2004 | A1 |
| 20070005349 | Absar et al. | Jan 2007 | A1 |
| 20080082325 | Tsuji et al. | Apr 2008 | A1 |
| Number | Date | Country |
|---|---|---|
| 4229372 | Mar 1994 | DE |
| 0169593 | Sep 2001 | WO |
| 0189086 | Nov 2001 | WO |
| 2004051627 | Jun 2004 | WO |
| Number | Date | Country | |
|---|---|---|---|
| 20070100639 A1 | May 2007 | US |
