HK40103944A - Method and device for unified time-domain / frequency domain coding of a sound signal - Google Patents

Description

Methods and apparatus for unified time-domain/frequency-domain coding of audio signals

Technical Field

This disclosure relates to a unified time-domain/frequency-domain coding apparatus and method for encoding input audio signals using a hybrid time-domain and frequency-domain coding mode, as well as a corresponding decoder apparatus and decoding method.

In this disclosure and the appended claims:

The term "sound" can refer to speech, general audio signals such as music and reverberant speech, and any other sound.

Background Technology

Existing session codecs can represent clean speech signals at bit rates of approximately 8 kbps with very good quality and are nearly transparent at bit rates of 16 kbps. However, at bit rates below 16 kbps, low-processing-latency session codecs (which most often encode input speech signals in the time domain) are unsuitable for general audio signals such as music and reverberant speech. To overcome this shortcoming, switching codecs have been introduced, essentially using a time-domain approach to encode speech-dominant input sound signals and a frequency-domain approach to encode general audio signals. However, this switching solution typically requires longer processing latency due to speech-music classification and the computation of the frequency-domain transform.

To overcome the aforementioned drawbacks associated with longer processing latency, a more unified time-domain and frequency-domain coding model has been proposed in U.S. Patent No. 9,015,038 (see Reference [1], the entire contents of which are incorporated herein by reference). This unified time-domain and frequency-domain coding model is part of the EVS (Enhanced Voice Services) sound codec standardized by 3GPP (3rd Generation Partnership Project) as described in Reference [2], the entire contents of which are incorporated herein by reference. In recent years, 3GPP has begun working on developing a 3D (three-dimensional) sound codec based on the EVS codec for immersive services known as IVAS (Immersive Voice and Audio Services) (see Reference [3], the entire contents of which are incorporated herein by reference).

To make the coding model even more effective for certain types of signals, coding schemes have been added to efficiently allocate available bits between the time and frequency domains and between low and high frequencies. The additional coding schemes are triggered by a new speech/music classifier whose output allows for ambiguous categories of signals that cannot be clearly classified as music or speech (see reference [4], the entire contents of which are incorporated herein by reference).

Summary of the Invention

This disclosure relates to a unified time-domain/frequency-domain coding method for encoding an input audio signal. The method includes: classifying the input audio signal into one of a plurality of audio signal categories, wherein the audio signal categories include an ambiguous signal type category indicating that the nature of the input audio signal is unclear; selecting one of a plurality of coding sub-modes for encoding the input audio signal when it is classified into the ambiguous signal type category; and performing hybrid time-domain/frequency-domain coding on the input audio signal using the selected coding sub-mode.

This disclosure also relates to a unified time-domain/frequency-domain coding method for encoding an input audio signal, comprising: classifying the input audio signal into one of a plurality of audio signal categories, wherein the audio signal categories include an ambiguous signal type category indicating that the nature of the input audio signal is unclear; and performing hybrid time-domain/frequency-domain coding on the input audio signal in response to the input audio signal being classified into the ambiguous signal type category. Hybrid time-domain/frequency-domain coding of the input audio signal includes band selection and bit allocation, for selecting a frequency band to be quantized and for allocating a bit budget available for quantization among the selected frequency bands.

According to this disclosure, a unified time-domain/frequency-domain coding apparatus for encoding an input sound signal is also provided, comprising: a classifier for classifying the input sound signal into one of a plurality of sound signal categories, wherein the sound signal categories include an ambiguous signal type category indicating that the nature of the input sound signal is unclear; a selector for encoding one of a plurality of coding sub-modes when the input sound signal is classified into the ambiguous signal type category; and a hybrid time-domain/frequency-domain encoder for encoding the input sound signal using the selected coding sub-mode.

This disclosure also relates to a unified time-domain/frequency-domain coding apparatus for encoding an input audio signal, comprising: a classifier for classifying the input audio signal into one of a plurality of audio signal categories, wherein the audio signal categories include an ambiguous signal type category indicating that the nature of the input audio signal is unclear; and a hybrid time-domain/frequency-domain encoder for encoding the input audio signal in response to the input audio signal being classified into an ambiguous signal type category. The hybrid time-domain/frequency-domain encoder includes a frequency band selector and a bit allocator for selecting a frequency band to be quantized and for allocating a bit budget available for quantization among the selected frequency bands.

This invention provides a method for decoding a sound signal, comprising: receiving a bitstream conveying information for reconstructing a mixed time-domain/frequency-domain excitation, the mixed time-domain/frequency-domain excitation representing a sound signal classified as an unclear signal type category, the unclear signal type category indicating that the nature of the sound signal is unclear, wherein the information includes one of a plurality of coding sub-patterns for encoding the sound signal classified as an unclear signal type category; reconstructing the mixed time-domain/frequency-domain excitation in response to the information including the coding sub-patterns for encoding the input sound signal transmitted in the bitstream; converting the mixed time-domain/frequency-domain excitation to the time domain; and filtering the converted mixed time-domain/frequency-domain excitation to the time domain using a synthesis filter to generate a synthesized version of the sound signal.

This disclosure proposes a method for decoding an audio signal, comprising: receiving a bitstream conveying information available for reconstructing a hybrid time-domain/frequency-domain excitation, the hybrid time-domain/frequency-domain excitation representing an audio signal, the audio signal being (a) classified into an ambiguous signal type category, the ambiguous signal type category indicating that the nature of the audio signal is ambiguous, and (b) encoding it using (i) a frequency band selected for quantization and (ii) a bit budget available for quantization allocated between the frequency bands; reconstructing the hybrid time-domain/frequency-domain excitation in response to the information transmitted in the bitstream, wherein reconstructing the hybrid time-domain/frequency-domain excitation includes selecting the frequency band for quantization and allocating the bit budget available for quantization between the frequency bands; converting the hybrid time-domain/frequency-domain excitation to the time domain; and filtering the converted hybrid time-domain/frequency-domain excitation to the time domain using a synthesis filter to produce a synthesized version of the audio signal.

According to this disclosure, a sound signal decoder is provided, comprising: a receiver of a bitstream transmitting information for reconstructing a mixed time-domain/frequency-domain excitation, the mixed time-domain/frequency-domain excitation representing a sound signal classified as an unclear signal type category, the unclear signal type category indicating that the nature of the sound signal is unclear, wherein the information includes one of a plurality of coding sub-patterns for encoding the sound signal classified as an unclear signal type category; a reconstructor responsive to the information in the bitstream including the coding sub-patterns for encoding an input sound signal; a converter for converting the mixed time-domain/frequency-domain excitation to the time domain; and a synthesis filter for filtering the converted mixed time-domain/frequency-domain excitation to the time domain to produce a synthesized version of the sound signal.

This disclosure also relates to an audio signal decoder, comprising: a receiver of a bitstream transmitting information for reconstructing a mixed time-domain/frequency-domain excitation, the mixed time-domain/frequency-domain excitation representing an audio signal, the audio signal being (a) classified into an ambiguous signal type category, the ambiguous signal type category indicating that the nature of the audio signal is ambiguous, and (b) encoded using (i) a frequency band selected for quantization and (ii) a bit budget allocated between the frequency bands for quantization; a reconstructor of the mixed time-domain/frequency-domain excitation in response to the information transmitted in the bitstream, wherein the reconstructor selects the frequency band for quantization and the allocation of the bit budget available for quantization between the frequency bands; a converter for converting the mixed time-domain/frequency-domain excitation to the time domain; and a synthesis filter for filtering the converted mixed time-domain/frequency-domain excitation to the time domain to produce a synthesized version of the audio signal.

The foregoing and other features will become more apparent from the following non-limiting description of illustrative embodiments of a unified time-domain/frequency-domain coding method, a unified time-domain/frequency-domain coding apparatus, a decoding method, and a decoder apparatus, given only as examples with reference to the accompanying drawings.

Attached Figure Description

In the attached diagram:

Figure 1 is a schematic block diagram that simultaneously illustrates the unified time-domain/frequency-domain CELP (code-excited linear prediction) coding method and the corresponding unified time-domain/frequency-domain CELP coding device (e.g., ACELP (algebraic code-excited linear prediction) coding method and device).

Figure 2 is a schematic block diagram of a more detailed structure of the unified time-domain/frequency-domain coding method and device of Figure 1, wherein the preprocessor performs first-level analysis to classify the input audio signal;

Figure 3 is a schematic block diagram that simultaneously shows the calculator for the cut-off frequency of the time-domain excitation contribution and the corresponding operation for estimating the cut-off frequency;

Figure 4 is a schematic block diagram showing the calculator for the cutoff frequency of Figure 3 and the corresponding operation for estimating the cutoff frequency in more detail.

Figure 5 is a schematic block diagram that simultaneously shows an overview of the frequency quantizer and the corresponding frequency quantization operation;

Figure 6 is a schematic block diagram of the more detailed structure of the frequency quantizer and frequency quantization operation in Figure 5;

Figure 7 is a schematic block diagram showing both the unified time-domain/frequency-domain CELP coding method and the corresponding alternative implementations of the unified time-domain/frequency-domain CELP coding device;

Figure 8 is a schematic block diagram showing the operation of selecting an encoding sub-mode and the corresponding sub-mode selector;

Figure 9 is a schematic block diagram showing the band selector and bit allocator, as well as the corresponding operations of band selection and bit allocation, which is used to allocate the available bit budget to the frequency domain coding mode when the input audio signal is neither classified as speech nor music in the alternative implementations of Figures 7 and 8.

Figure 10 is a simplified block diagram of an example configuration of hardware components for forming a unified time-domain/frequency-domain coding device and method for encoding input audio signals;

Figure 11 is a schematic block diagram showing a decoder device 1100 and a corresponding decoding method 1150 for decoding bitstreams from the unified time-domain/frequency-domain coding device and the corresponding unified time-domain/frequency-domain coding method of Figure 7; and

Figure 12 is a schematic block diagram showing both a sound signal decoder and a corresponding sound signal decoding method. The sound signal decoder and the corresponding sound signal decoding method are used to decode bitstreams from a unified time-domain/frequency-domain coding device and a corresponding unified time-domain/frequency-domain coding method when the sound signal is classified as having an unclear signal type category.

Detailed Implementation

This disclosure proposes a unified time-domain and frequency-domain coding model that improves the synthesis quality of general audio signals (e.g., music and/or reverberant speech) without increasing processing latency and bit rate. The unified time-domain and frequency-domain coding model includes:

- A time-domain coding scheme operating in the linear prediction (LP) residual domain, where available bits are dynamically allocated among an adaptive codebook, one or more fixed codebooks (e.g., algebraic codebooks, Gaussian codebooks, etc.), and a variable-length fixed codebook; and

-Frequency domain coding mode

It depends on the characteristics of the input sound signal.

To achieve low processing latency and low bit rate session sound codecs (which improve the synthesis quality of general audio signals such as music and/or reverberant speech), the frequency domain coding scheme is integrated as closely as possible to the CELP (Code-Excited Linear Prediction) time domain coding scheme. For this purpose, the frequency domain coding scheme uses a frequency transformation performed in the LP (Linear Prediction) residual domain. This allows switching from one frame (e.g., a 20ms frame) to another with almost no artifacts. As is well known in the field of sound codecs, the input audio signal is sampled at a given sampling rate and processed by groups of these samples called “frames,” which are typically divided into multiple “subframes.” Here, the integrals of the two (2) time-domain and frequency-domain coding schemes are close enough to allow the bit budget to be dynamically reallocated to another coding scheme if it is determined that the current coding scheme is not efficient enough.

A feature of the proposed unified time-domain and frequency-domain coding model is the variable-time support for the time-domain components, which varies from quarter-frames (subframes) to full frames on a frame-by-frame basis. As a non-limiting illustrative example, a frame can represent a 20ms input audio signal. If the audio codec's internal sampling rate is 16kHz, such a frame corresponds to 320 samples of the input audio signal, or if the codec's internal sampling rate is 12.8kHz, such a frame corresponds to 256 samples per frame. Subframes (in this example, a quarter of a frame) then represent either 80 or 64 samples, depending on the audio codec's internal sampling rate. In this non-limiting illustrative embodiment, the audio codec's internal sampling rate is 12.8kHz, which gives an input audio signal with a frame length of 256 samples and a subframe length of 64 samples.

Variable timing support allows for the capture of major temporal events at a minimum bit rate to create a basic temporal excitation contribution. At very low bit rates, the timing support is typically the entire frame. In this case, the temporal contribution of the excitation consists only of an adaptive codebook; then, the corresponding adaptive codebook (pitch) information and gain are transmitted once per frame. When more bit rate is available, more temporal events can be captured by shortening the timing support and increasing the bit rate allocated to the temporal coding mode. Finally, when the timing support is sufficiently short (less than a quarter of a frame (subframe)) and the available bit rate is sufficiently high, for each subframe, the temporal contribution of the excitation can include an adaptive codebook contribution with a corresponding adaptive codebook gain, a fixed codebook contribution with a corresponding fixed codebook gain, or both an adaptive codebook contribution with a corresponding gain and a fixed codebook contribution. Optionally, for each half of a frame (subframe), an adaptive codebook contribution with a corresponding adaptive codebook gain and a fixed codebook contribution with a corresponding fixed codebook gain can also be transmitted; this has the advantage of not consuming too much bit rate while still being able to encode temporal events. Parameters describing the codebook index and gain are then transmitted for each subframe.

At low bit rates, conversational sound codecs cannot adequately encode higher frequencies. This leads to a significant reduction in synthesis quality when the input audio signal includes music and/or reverberant speech. To address this issue, a feature is added to calculate the efficiency of the temporal excitation contribution. In some cases, the temporal excitation contribution is not valuable regardless of the input bit rate and time frame support. In these cases, all bits are reallocated to the next step of frequency domain encoding. However, most of the time, the temporal excitation contribution is only valuable at a certain frequency (hereinafter referred to as the "cutoff frequency"). In these cases, the temporal excitation contribution is filtered above the cutoff frequency. The filtering operation allows valuable information encoded with the temporal excitation contribution to be preserved, while worthless information above the cutoff frequency is removed. In a non-limiting illustrative embodiment, filtering is performed in the frequency domain by setting the frequency bin above a certain frequency (cutoff frequency) to zero.

The combination of variable time support and variable cutoff frequency makes bit allocation within the unified time-domain and frequency-domain coding model highly dynamic. The bit rate after quantization of the LP filter can be allocated entirely to the time domain, entirely to the frequency domain, or somewhere in between. The bit rate allocation between the time and frequency domains is a function of the number of subframes contributed to the time-domain excitation, the available bit budget, and the calculated cutoff frequency. To make the unified time-domain and frequency-domain coding model even more effective for certain types of input audio signals, specific coding sub-patterns are added to efficiently allocate available bits between the time and frequency domains, and between low and high frequencies. These added specific coding sub-patterns are determined using a novel speech/music audio classifier that produces an output that allows for signals of unclear category (signals that cannot be clearly classified as music or speech).

To create a total excitation that more effectively matches the input LP residuals, a frequency-domain coding mode is applied. One feature is performing frequency-domain coding on a vector containing the difference between the frequency representation (frequency transform) of the input LP residuals and the frequency representation (frequency transform) of the filtered temporal excitation contribution up to the cutoff frequency, and also containing the frequency representation (frequency transform) of the input LP residuals themselves above that cutoff frequency. A smooth spectral transition is inserted only between the two segments above the cutoff frequency. In other words, the high-frequency portion of the frequency representation of the temporal excitation contribution first zeros out above the cutoff frequency. A transition region between the unchanged portion of the spectrum of the temporal excitation contribution and the zero-out portion of the spectrum is inserted just above the cutoff frequency to ensure a smooth transition between the two portions of the spectrum. This modified spectrum of the temporal excitation contribution is then subtracted from the frequency representation of the input LP residuals. Therefore, the resulting spectrum corresponds to the difference between the two spectra below the cutoff frequency and to the frequency representation of the LP residuals above the cutoff frequency, with some transition region. As mentioned above, the cutoff frequency can vary from one frame to another.

Regardless of the frequency quantization method (frequency domain coding mode) chosen, there is always the possibility of pre-echo, especially in the case of long windows. In the technique disclosed herein, the window used is a square window, such that the additional window length compared to the encoded input audio signal is zero (0), i.e., no overlap addition is used. While this corresponds to the optimal window for reducing any potential pre-echo, some pre-echo can still be audible under time attacks. Many techniques exist to address such pre-echo problems, but this disclosure proposes a simple feature for eliminating this pre-echo problem. This feature is based on a memoryless time domain coding mode derived from the “Transition Mode” of ITU-T Recommendation G.718; see reference [5], sections 6.8.1.4 and 6.8.4.2, the entire contents of which are incorporated herein by reference. The idea behind this feature is to take advantage of the fact that the proposed unified time and frequency domain coding model is integrated into the LP residual domain, which allows for switching with almost no artifacts at any time. When the input audio signal is considered general audio (music and/or reverberant speech) and a temporal attack is detected in the frame, the frame is encoded using only a memoryless temporal coding scheme. This memoryless temporal coding scheme takes into account temporal attacks, thus avoiding pre-echoes that could be introduced when using frequency-domain coding of the frame.

Non-limiting illustrative examples

In the proposed unified time-domain and frequency-domain coding model, the aforementioned adaptive codebook, one or more fixed codebooks (e.g., algebraic codebooks, Gaussian codebooks, etc.) (i.e., the so-called time-domain codebook), and frequency-domain quantization (frequency-domain coding mode) can be considered as a codebook library, and bits can be allocated among all available codebooks or subsets thereof. This means that, for example, if the input audio signal is clean speech, all bits will be allocated to the time-domain coding mode, thus essentially reducing the coding to the traditional CELP scheme. On the other hand, for some musical segments, all bits allocated to encoding the input LP residual are sometimes best spent in the frequency domain, such as in the transform domain. Furthermore, specific cases can be added where (a) the time domain uses a larger portion of the total available bit rate to encode more time-domain events, while still reserving bits for encoding some frequency information, or (b) low-frequency content takes precedence over high-frequency content, and vice versa.

As indicated in the preceding description, the time support for time-domain and frequency-domain coding modes does not need to be the same. While the bits spent on different time-domain coding operations (adaptive and algebraic book search) are typically allocated based on subframes (usually a quarter of a frame, or 5ms of time support), the bits allocated to the frequency-domain coding mode are allocated based on frames (usually 20ms of time support) to improve frequency resolution.

The bit budget allocated to the time-domain CELP coding mode can also be dynamically controlled based on the input audio signal. In some cases, the bit budget allocated to the time-domain CELP coding mode can be zero, effectively meaning that the entire bit budget is attributed to the frequency-domain coding mode. The choice between the time-domain and frequency-domain coding modes operating in the LP residual domain has two (2) main benefits. First, it is compatible with the time-domain CELP coding mode, proving effective in speech signal coding. Therefore, no artifacts are introduced due to the switching between the two types of coding modes (time-domain and frequency-domain coding modes). Second, the lower dynamics of the LP residual relative to the original input audio signal and its relative flatness make it easier to use a square window for frequency transformation, thus allowing the use of a non-overlapping window.

In a non-restrictive example where the codec uses a rate of 12.8 kHz (meaning 256 samples per frame), similar to ITU-T Recommendation G.718 (Reference [5]), the length of the subframes used in the time-domain CELP coding mode can vary from a typical 1/4 (5 ms) of the frame length to half a frame (10 ms) or a full frame length (20 ms). The subframe length is determined based on the available bit rate and analysis of the input audio signal, particularly the spectral dynamics of the input audio signal. The subframe length can be determined in a closed-loop manner. To save complexity, the subframe length can also be determined in an open-loop manner. The subframe length can also be controlled by the properties of the input audio signal detected by a signal classifier (e.g., a speech/music classifier). The subframe length can be changed frame by frame.

Once the subframe length is selected in the current frame, standard closed-loop tone analysis is performed, and a first contribution to the excitation signal is selected from the adaptive codebook. Then, depending on the available bit budget and the characteristics of the input audio signal (e.g., in the case of an input speech signal), a second contribution from one or more fixed codebooks can be added before the transformation in the transform domain. The resulting excitation contribution is the time-domain excitation contribution. On the other hand, at very low bit rates and in the case of general audio signals, it is generally better to skip the fixed codebook stage and use all remaining bits for transform-domain coding. The transform-domain coding can be, for example, a frequency-domain coding mode. As mentioned above, the subframe length can be one-quarter, half, or one full frame. The fixed codebook contribution is used only when the subframe length is equal to one-quarter of the frame length. When the subframe length is determined to be half or the full frame length, only the adaptive codebook contribution is used to represent the time-domain excitation contribution, and all remaining bits are allocated to the frequency-domain coding mode. Alternatively, an additional coding mode will be described where the fixed codebook can be used when the subframe length is equal to half the frame length. This addition is made to improve the quality of specific types of input audio signals that contain time events, while maintaining an acceptable bit budget to encode the frequency domain excitation contribution.

Once the time-domain excitation contribution has been calculated, its efficiency needs to be evaluated and quantified. If the gain of coding in the time domain is very low, it is more efficient to completely remove the time-domain excitation contribution and use all bits for the frequency-domain coding mode. On the other hand, for example, in the case of a clean input speech signal, a frequency-domain coding mode is not needed, and all bits are allocated to the time-domain coding mode. However, coding in the time domain is typically only effective at a certain frequency. This frequency corresponds to the aforementioned cutoff frequency of the time-domain excitation contribution. Determining this cutoff frequency ensures that the entire time-domain coding contributes to a better final synthesis, rather than the opposite of frequency-domain coding.

The cutoff frequency can be estimated in the frequency domain. To calculate the cutoff frequency, the spectrum of the LP residual and the temporal excitation contribution is first divided into a predetermined number of frequency bands, with multiple frequency segments defined within each band. The number of frequency bands and the number of frequency segments covered by each band can vary from one implementation to another. For each frequency band, a normalized correlation is calculated between the frequency representation of the temporal excitation contribution and the frequency representation of the LP residual, and the correlation between adjacent frequency bands is smoothed. As a non-limiting example, the lower bound for the correlation per band is 0.5 and normalized between 0 and 1, and then the average correlation is calculated as the average of the correlations across all frequency bands. To estimate the cutoff frequency initially, the average correlation is then scaled between 0 and half of the internal sampling rate (half of the internal sampling rate corresponds to a normalized correlation value of 1). At very low bit rates or for additional coded sub-modes as described below, the average correlation is doubled before finding the cutoff frequency. This is done when the temporal excitation contribution is known to be needed even if the correlation is not very high due to the use of a low bit rate or because the type of input audio signal does not allow for high correlation. The first estimate of the cutoff frequency is then found as the upper bound of the band that is closest to the value of the scaled average correlation. In the implemented example, sixteen (16) bands at an internal sampling rate of 12.8 kHz are defined for correlation calculation.

Leveraging the psychoacoustic properties of the human ear, the reliability of the cutoff frequency estimation can be improved by comparing the estimated position of the eighth harmonic frequency of the pitch with the cutoff frequency estimated through correlation calculations. If the position is higher than the cutoff frequency estimated through correlation calculations, the cutoff frequency is modified to correspond to the position of the eighth harmonic frequency of the pitch. If one of the additional coding sub-modes is used, the cutoff frequency has a minimum value higher than or equal to, for example, 2775 Hz (7th band). The final value of the cutoff frequency is then quantized and sent to a remote decoder. In the implemented example, 3 or 4 bits are used for this quantization, thus giving 8 or 16 possible cutoff frequencies depending on the bit rate.

Once the cutoff frequency is known, frequency quantization of the frequency-domain excitation contribution is performed. First, the difference between the frequency representation (frequency transform) of the input LP residual and the frequency representation (frequency transform) of the time-domain excitation contribution is determined. Then, a new vector is created that includes this difference up to the cutoff frequency, as well as a smooth transition to the frequency representation of the input LP residual to the remaining spectrum. Frequency quantization is then applied to the entire new vector. In the implemented example, quantization involves encoding the sign and position of the dominant (most energetic) spectral pulses. The number of pulses to be quantized per frequency band is related to the bit rate available for the frequency-domain coding mode. If the available bits are insufficient to cover all frequency bands, the remaining bands are simply filled with noise.

Frequency quantization of a band using the quantization methods described in the preceding paragraphs does not guarantee that all frequency bands within that band will be quantized. This is especially true at low bit rates where the number of quantized spectral pulses per band is relatively low. To prevent audible artifacts caused by these unquantized frequency bands, some noise is added to fill these gaps. Since at low bit rates, the quantized spectral pulses should dominate the spectrum rather than the inserted noise, the noise spectral amplitude corresponds only to a portion of the pulse amplitude. When the available bit budget is low (allowing for more noise), the amplitude of the noise added to the spectrum is higher, and when the available bit budget is high, the amplitude of the noise added to the spectrum is lower.

In frequency-domain coding mode, a gain is calculated for each frequency band to match the energy of the unquantized signal with that of the quantized signal. The gain is vector-quantized and applied band-by-band to the quantized signal. For example, when a unified time-domain and frequency-domain coding model changes bit allocation from a time-only coding mode to a hybrid time-domain/frequency-domain coding mode, the per-band excitation spectral energy in the time-only coding mode does not match the per-band excitation spectral energy in the hybrid time-domain/frequency-domain coding mode. This energy mismatch can produce some switching artifacts, especially at low bit rates. To reduce any audible degradation caused by this bit reallocation, a long-term gain can be calculated for each frequency band, and this long-term gain can be applied to correct the energy of each frequency band for several frames after switching from a time-only coding mode to a hybrid time-domain/frequency-domain coding mode.

After completing the frequency domain coding pattern, the total excitation is found by adding the frequency representations (frequency transformations) of the frequency domain excitation contribution and the time domain excitation contribution. Then, the sum of these two (2) excitation contributions is transformed back to the time domain to form the total excitation. Finally, the total excitation is filtered by an LP synthesis filter to calculate the synthesized signal.

In one embodiment, although only temporal excitation contributions are used to update the CELP-coded memory on a subframe basis, the total excitation is used to update those memories at frame boundaries.

In another possible implementation, the CELP-encoded memory is updated at frame boundaries on a subframe basis, using only temporal excitation contributions. This results in an embedded architecture where the frequency-domain coded signal constitutes an upquantization layer independent of the core CELP layer. In this particular case, a fixed codebook is always used to update the adaptive codebook content. However, the frequency-domain coding mode can be applied to the entire frame. This embedded approach is suitable for bit rates of approximately 12 kbps and higher.

1) Classification of sound signal types

Figure 1 is a schematic block diagram illustrating an overview of a unified time-domain/frequency-domain CELP coding method 150 and a corresponding unified time-domain/frequency-domain CELP coding device 100 (e.g., ACELP method and device). Of course, other types of CELP coding methods and devices can be implemented using the same concepts.

Figure 2 is a schematic block diagram of a more detailed structure of the unified time-domain/frequency-domain CELP coding method 150 and device 100 of Figure 1.

The unified time-domain/frequency-domain CELP coding apparatus 100 includes a preprocessor 102 (FIG. 1) for performing operations 152 of analyzing parameters of an input audio signal 101 (FIG. 1 and 2). Referring to FIG. 2, the preprocessor 102 includes: an LP analyzer 201 for performing LP analysis of the input audio signal 101; a spectrum analyzer 202 for performing spectrum analysis; an open-loop tone analyzer 203 for performing open-loop tone analysis; and a signal classifier 204 for performing classification of the input audio signal. Analyzers 201 and 202 and associated operations 251 and 252 perform LP and spectrum analysis typically performed in CELP coding, as described, for example, in ITU-T Recommendation G.718, Reference [5], sections 6.4 and 6.1.4, and therefore will not be described further in this disclosure.

The preprocessor 102 performs a first-level analysis to classify the input sound signal 101 between speech and non-speech (general audio (music or reverberant speech)), for example in a manner similar to that described in reference [6], the entire contents of which are incorporated herein by reference, or by utilizing any other reliable speech/non-speech differentiation method.

Following this first-level analysis, the preprocessor 102 performs a second-level analysis of the input signal parameters to allow time-domain CELP coding (no-frequency-domain coding) to be used for some audio signals that have strong non-speech characteristics but are still better coded using time-domain methods. When a significant energy change occurs, this second-level analysis allows the unified time-domain/frequency-domain CELP coding device 100 to switch to a memoryless time-domain coding mode, commonly referred to as the switching mode in reference [7], the entire contents of which are incorporated herein by reference.

During this second-level analysis, signal classifier 204 calculates and uses the smoothed version of the open-loop pitch-related change C _{_st}σ_{_C} from open-loop pitch analyzer 203, the current total frame energy E_{_tot} (the total energy of the input sound signal in the current frame), and the difference E _{_diff} between the current total frame energy and the previous total frame energy. First, signal classifier 204 calculates the smoothed open-loop pitch-related change using, for example, the following relationship:

in:

-C _st is a smooth open-loop pitch correlation defined as follows: C _st = 0.9·C _cl + 0.1·C _st ;

-C _ol is the open-loop tone correlation calculated by analyzer 203 using methods known to those skilled in the art of CELP coding, for example, as in ITU-T Recommendation G.718, references

[5], as described in Section 6.6;

- is the average of the last 10 frames i of the smooth open-loop pitch correlation C _st ;

-σ _C is a smooth, open-loop pitch-related variation.

When the signal classifier 204 classifies a frame as non-speech during the first-level analysis, it performs the following verification to determine in the second-level analysis whether using the hybrid time-domain/frequency-domain coding mode is indeed safe. However, sometimes it is better to encode the current frame using only the time-domain coding mode, using one of the time-domain methods estimated by the preprocessing function of the time-domain coding mode. Specifically, it may be better to use a memoryless time-domain symbol mode to minimize any possible pre-echoes that can be introduced by the hybrid time-domain/frequency-domain coding mode.

As a non-limiting implementation for the first verification of whether a hybrid time-domain/frequency-domain coding mode should be used, the signal classifier 204 calculates the difference between the current total frame energy and the previous frame total energy. When the difference between the current total frame energy E _{_tot} and the previous frame total energy E _{_diff} is higher than, for example, 6 dB, this corresponds to a so-called “time attack” in the input audio signal 101. In this case, the speech/non-speech determination and the selected coding mode are rewritten, and a memoryless time-domain coding mode is enforced. More specifically, the unified time-domain/frequency-domain CELP coding device 100 includes a time/time-frequency coding selector 103 (FIG. 1) for performing an operation 153 to select between time-domain-only coding and hybrid time-domain/frequency-domain coding. To this end, the time/time-frequency coding selector 103 includes: a speech/general audio selector 205 (FIG. 2) for performing an operation 255 of selecting between speech and general audio to classify the input sound signal 101; a time attack detector 208 (FIG. 2) for performing an operation 258 of detecting time attacks in the input sound signal 101; and a selector 206 (FIG. 2) for performing an operation 256 of selecting a memoryless time-domain coding mode. In other words:

- In response to the selection of the speech signal by the selector 205, the operation 257 of CELP encoding of the speech signal is performed using the closed-loop CELP encoder 207 (Figure 2).

In response to the selection of non-speech signals (general audio) by selector 205 and the detection of time attacks in input audio signal 101 by detector 208, selector 206 forces closed-loop CELP encoder 207 (FIG. 2) to use memoryless time-domain coding mode to encode the input audio signal.

The closed-loop CELP encoder 207 forms a part of the time-domain encoder 104 of FIG1. Closed-loop CELP encoders are well known to those skilled in the art and will not be described further in this specification.

As a non-limiting implementation for the second verification of whether a hybrid time-domain/frequency-domain coding mode should be used, when the difference between the current total frame energy _Etot and the previous frame total energy _Ediff is less than or equal to 6dB, but:

- Smooth open-loop pitch correlation C _st greater than 0.96; or

- Smooth open-loop pitch correlation C _{_st} is greater than 0.85, and the difference between the current total frame energy E _{_tot} and the previous frame total energy E _{_diff} is less than 0.3 dB; or

- The change in smooth open-loop pitch correlation _σC is less than 0.1, and the difference between the current total frame energy _Etot and the total energy of the last previous frame _Ediff is less than 0.6 dB; or

- The current total frame energy E _tot is less than 20 dB;

And this is at least the second consecutive frame (cnt≥2), in which the decision of the first-level analysis is changed, and then the speech/general audio selector 205 determines that the current frame will be encoded using the closed-loop CELP encoder 207 (Figure 2) in time-only coding mode.

Otherwise, the time/time-frequency coding selector 103 selects a hybrid time-domain/frequency-domain coding mode, as disclosed in the following description.

For example, when the non-voice input sound signal is music, the following pseudocode can be used to summarize the second verification:

Where E _tot is the current total frame energy, expressed as:

Where x(i) represents the sample of the input audio signal in the current frame, N is the number of samples of the input audio signal per frame, and E _diff is the difference between the total energy of the current frame E _tot and the total energy of the last previous frame.

Figure 7 is a schematic block diagram showing both the unified time-domain/frequency-domain CELP coding method 750 and the corresponding unified time-domain/frequency-domain CELP coding device 700, wherein the preprocessor 702 also performs a first-level analysis to classify the input audio signal 101.

Specifically, the unified time-domain/frequency-domain CELP coding method 750 includes an operation 752 as described in reference [4], preprocessing the input audio signal 101 to obtain parameters required for classifying the input audio signal. To perform operation 752, the hybrid time-domain/frequency-domain CELP coding device 700 includes a preprocessor 702.

The Unified Time/Frequency CELP Coding Method 750 includes an operation 751 that classifies the input audio signal 101 into speech, music, and unclear signal type categories using parameters from the preprocessor 702 in a manner similar to that described in reference [4], or using any other reliable speech/music and unclear signal type differentiation method. An unclear signal type category indicates that the nature of the input audio signal 101 is unclear, and specifically, the input audio signal 101 is neither classified as speech nor as music. To perform operation 751, the Unified Time/Frequency CELP Coding Apparatus 700 includes an audio signal classifier 701.

If the sound signal classifier 701 classifies the input sound signal 101 into a music category, the frequency domain encoder 703 performs an operation 753 to encode the input sound signal 101 using frequency domain coding as described, for example, in reference [2]. The frequency domain-coded music signal can then be synthesized in a music synthesis operation 754 performed by the synthesizer 704 to recover the music signal.

Similarly, if the sound signal classifier 701 classifies the input sound signal 101 into a speech category, the temporal encoder 705 performs an operation 755 to encode the input sound signal 101 using temporal coding as described, for example, in reference [2]. The temporally encoded speech signal can then be synthesized in a synthesis filtering operation 756 performed by the synthesizer 706 to recover the speech signal.

Therefore, the unified time-domain/frequency-domain coding apparatus 700 and method 750 maximize the performance of time-domain-only coding and frequency-domain-only coding by limiting their use to input audio signals with clear speech characteristics and input audio signals with clear musical characteristics, respectively. This improves the overall quality of all types of input audio signals from low to medium bit rates.

The coding sub-modes have been designed as part of the unified time-domain and frequency-domain coding model to efficiently encode input sound signals that are not classified as speech or music (the signal type category is unclear). Two (2) bits are used to transmit three (3) coding sub-modes identified by the corresponding sub-mode flags. The fourth sub-mode allows backward interoperability with the traditional unified time-domain and frequency-domain coding model (EVS).

As shown in Figure 8, the operation 751 of classifying the input audio signal 101 includes an operation 850 of selecting one of the encoding sub-modes in response to the bit rate available for encoding the input audio signal 101 and the characteristics of the input audio signal that are classified as having an unclear signal type category. To perform operation 850, the audio signal classifier 701 includes a sub-mode selector 800.

The encoding sub-pattern is identified by the sub-pattern flag F _tfsm . In the non-restrictive implementation of Figure 8, the sub-pattern selector 800 selects the encoding sub-pattern as follows:

If (a) the bit rate available for encoding the input audio signal 101 is no higher than 9.2 kbps and (b) the input audio signal 101 is neither classified as speech nor as music, then the sub-mode selector 800 selects the aforementioned backward coding sub-mode (see 803). The sub-mode flag F _tfsm is then set to "0" (see 802). The selection of the backward coding mode results in the use of the traditional unified time-domain and frequency-domain coding model of Figures 1 and 2 (EV).

- If (a) the input audio signal 101 is not classified as speech or music by classifier 701, and the available bit rate is high enough to allow for adaptive and fixed codebook encoding and gain, typically meaning a bit rate higher than 9.2 kbps (see 803), (b) the probability that the input audio signal 101 is music (a weighted speech/music decision tending towards music, wdlp(n)) is not greater than “0” (see 804), and (c) there is no possibility of a time attack being detected in the current frame of the input audio signal (the transition counter is not greater than “0”, as described in ITU-T Recommendation G.718, Reference [5], Sections 6.8.1.4 and 6.8.4.2) (see 806), then the submode selector 800 selects the first encoding submode. The submode flag F _tfsm is then set to “1” (see 801). Although the input sound signal 101 was not classified as speech or music by the classifier 701, the selector 800 detected speech-like characteristics in the input sound signal 101 and selected the first encoding sub-mode (sub-mode flag F _tfsm = 1) because CELP is not optimal for encoding this type of sound signal.

- If (a) the input audio signal 101 is neither classified as speech nor as music by classifier 701, and the available bit rate is high enough to allow for encoding and gain for both adaptive and fixed codebooks, typically meaning a bit rate greater than 9.2 kbps (see 803), (b) the probability that the input audio signal 101 is music (a weighted speech/music decision tending towards music, wdlp(n)) is not greater than “0” (see 804), and (c) the probability of detecting a time attack in the current frame of the input audio signal (a transition counter greater than “0”, as described in ITU-T Recommendation G.718, Reference [5], Sections 6.8.1.4 and 6.8.4.2) (see 806), then the submode selector 800 selects the second coding submode. The submode flag F _tfsm is then set to “2” (see 807). As will be explained in the following description, the second coded sub-mode (sub-mode flag F _tfsm = 2) allocates more bits to the lower portion of the spectrum.

If (a) the input audio signal 101 is neither classified as speech nor music by classifier 701, and the available bit rate is high enough to allow for at least adaptive codebook encoding and gain, and still has a large number of bits for frequency encoding, typically meaning a bit rate higher than 9.2 kbps (see 803), and (b) the probability that the input audio signal 101 is music (a weighted speech/music decision tending towards music, wdlp(n)) is greater than “0” (see 804), then submode selector 800 selects the third encoding submode. The submode flag F _tfsm is then set to “3” (see 808). Although the input audio signal 101 is neither classified as speech nor music by classifier 701, selector 800 detects “musical”-like characteristics in the input audio signal 101 and selects the third encoding submode (submode flag F _tfsm = 3). Such audio signal segments are still considered non-musical, but the sub-mode flag F _tfsm is set to "3" (selecting the third encoded sub-mode), indicating that the sample includes high-frequency or tonal content.

Reference [4] describes the probability that the input sound signal 101 is speech, music, or something in between. When the classification of speech or music is unclear, if the probability wdlp(n) is greater than 0, the signal is considered to have some musical characteristics. The following shows the threshold where the probability will be high enough to be considered as music or speech.

Table 1: Probability thresholds for categories that are unclear

The selected encoding sub-mode (e.g., sub-mode flag F _{_tfsm} ) is sent to the bitstream to the remote decoder. The path selected within the decoder depends on the signaling bits included in the bitstream. Once the decoder detects the presence of a frame encoded using hybrid time/frequency domain coding, it decodes the sub-mode flag F _{_tfsm} from the bitstream. If the detected sub-mode flag F _{_tfsm} is “0”, then the EVS backward interoperable traditional unified time and frequency domain coding model will be used to decode the remainder of the bitstream. On the other hand, if the sub-mode flag F _{_tfsm} is not “0”, sub-mode decoding is followed. The decoder will replicate the process followed by the encoder, particularly the bit allocation between the time and frequency domains and the bit allocation in different frequency bands, as described later in Section 6.2.

2) Determination of subframe length

In a typical CELP, input audio signal samples are processed in frames of 10-30ms, and these frames are divided into subframes for adaptive and fixed codebook analysis. For example, a 20ms frame (256 samples at an internal sampling rate of 12.8kHz) can be used and divided into four 5ms subframes. Variable subframe length is a feature used to integrate the time and frequency domains into a single coding pattern. The subframe length can vary from a typical quarter of the frame length to half or the full frame length. Of course, it is possible to use an additional number of subframes (subframe length).

As shown in Figure 2, the parameter analysis operation 152 of the unified time-domain/frequency-domain CELP coding method 150 includes operation 259 for determining the high-spectral dynamics of the input audio signal 101 and operation 260 for calculating the number of subframes frame by frame. To perform operations 259 and 260, the preprocessor 102 of the unified time-domain/frequency-domain CELP coding device 100 includes a high-spectral dynamics analyzer 209 and a subframe count calculator 210, respectively.

The determination of the length (number of subframes) or time support is made by calculator 210 based on the available bit rate and analysis of the input audio signal, particularly the high-spectral dynamics of the input audio signal 101 from analyzer 209 and the open-loop pitch analysis including the smooth open-loop pitch correlation C _st from analyzer 203. High-spectral dynamics analyzer 209 determines the high-spectral dynamics of the input audio signal 101 in response to information from spectrum analyzer 202. For example, as described in ITU-T Recommendation G.718, Reference [5], Section 6.7.2.2, the high-spectral dynamics are calculated as the input spectrum without a noise floor, thus giving a representation of the input spectral dynamics. The input audio signal 101 is no longer considered to have high-spectral dynamics when the average spectral dynamics of the input audio signal 101 in the band between 4.4 kHz and 6.4 kHz, as determined by analyzer 209, is less than, for example, 9.6 dB and the last frame is considered to have high-spectral dynamics. In this case, more bits can be allocated to frequencies below, for example, 4 kHz by adding more subframes to the time-domain coding mode or by forcing more pulses into the lower frequency portion of the frequency-domain coding mode.

On the other hand, if the average spectral dynamics of the input audio signal 101 increases by, for example, 4.5 dB relative to the average spectral dynamics of the last frame, which is not considered to have high spectral dynamics as determined by the analyzer 209, then the input audio signal 101 is considered to have high spectral dynamics content above, for example, 4 kHz. In this case, depending on the available bit rate, some additional bits are used to encode the high frequencies of the input audio signal 101 to allow one or more frequency pulses to be encoded.

The subframe length, as determined by calculator 210 (Figure 2), also depends on the bit budget available for encoding the input audio signal 101. At very low bit rates, such as below 9 kbps, only one subframe is available for time-domain coding; otherwise, the number of available bits will be insufficient for frequency-domain coding. At medium bit rates, such as between 9 kbps and 16 kbps, one subframe is used for cases where high-frequency content is included, and two subframes are used if no such content is included. For medium-high bit rates, such as around 16 kbps and higher, a four (4) subframe case also becomes available if the smooth open-loop tone correlation Cst defined above is above, for example, 0.8.

While the case with one or two subframes limits temporal coding to only adaptive codebook contributions (with encoded tone lag and tone gain), i.e., no fixed codebook is used in this case, the case with four (4) subframes allows for both adaptive and fixed codebook contributions if the available bit budget is sufficient. The case with four (4) subframes is allowed at bit rates starting from approximately 16 kbps upwards. Due to bit budget constraints, the temporal excitation contribution consists only of the adaptive codebook contribution at the lower bit rates. The fixed codebook contribution can be added at higher bit rates, for example, starting at 24 kbps. For all cases, the temporal coding efficiency will then be evaluated to determine at which frequency (the aforementioned cutoff frequency) such temporal coding is valuable.

When the input audio signal 101 is classified as an ambiguous signal type by the classifier 701 and the sub-mode flag F _tfsm is greater than zero "0", the alternative implementations of Figures 7 and 8 use the first, second, or third encoding sub-mode defined above.

The audio signal classifier 701 determines that the number of subframes is four (4) unless the sub-mode flag F _tfsm is set to "1" or "2" (selecting the first or second encoding sub-mode), which means that the content of the input audio signal 101 is closer to speech (the possibility of detecting speech-like features or time attacks in the input audio signal 101) and the available bit rate is less than 15 kbps. Specifically:

- In the first or second encoding sub-mode (with the sub-mode flag F _tfsm set to "1" or "2"), the audio signal classifier 701 determines the number of four (4) subframes unless the available bit rate for encoding the input audio signal 101 is less than 15 kbps; then, an encoding mode using two (2) subframes is selected. In both cases, the corresponding number of fixed codebooks is used, i.e., two (2) or four (4) fixed codebooks; and

- In the third encoding mode (the sub-mode flag F _tfsm is set to 3, which means that the content of the input audio signal 101 is closer to music (a “musical”-like feature is detected in the input audio signal 101)), the audio signal classifier 701 determines the number of subframes to be four (4), but there is no fixed codebook contribution to keep more bits available for frequency domain excitation contribution unless the available bit rate for encoding the input audio signal 101 is greater than or equal to 22.6 kbps.

3) Closed-loop pitch analysis

In the unified time-domain/frequency-domain CELP encoding device 100 and method 150 (FIG. 1), when the selector 205 selects general audio as the classification of the input audio signal 101 and no time attack is detected in the detector 208, the hybrid time-domain/frequency-domain encoding method 170 and the corresponding hybrid time-domain/frequency-domain encoder 120 are used. Alternatively, in the unified time-domain/frequency-domain CELP encoding device 700 and method 750 (FIG. 7), when the audio signal classifier 701 classifies the input audio signal 101 into the "unclear signal type" category and selects one of the first, second, and third encoding sub-modes defined above (the sub-mode flag F _tfsm is set to "1", "2", or "3"), the hybrid time-domain/frequency-domain encoding method 770 and the corresponding hybrid time-domain/frequency-domain encoder 720 are used.

When using a hybrid time/frequency coding mode, closed-loop tone analysis is performed, followed by a fixed algebra search if necessary. For this purpose, the hybrid time/frequency coding method 170/770 includes an operation 155 for calculating the temporal excitation contribution. To perform operation 155, the hybrid time/frequency encoder 120/720 includes a calculator 105 for the temporal excitation contribution. The calculator 105 itself includes an analyzer 211 (FIG. 2) that performs a closed-loop tone analysis operation 261 in response to the open-loop tone analysis performed in the open-loop tone analyzer 203 (or preprocessor 702) and the subframe length (or the number of subframes in a frame) determined in the calculator 210 or the sound signal classifier 701. Closed-loop tone analysis is well known to those skilled in the art and examples of its implementation are described, for example, in ITU-T Recommendation G.718, reference [5]; section 6.8.4.1.4.1. Closed-loop pitch analysis results in the computation of pitch parameters, also known as adaptive codebook parameters, which primarily consist of pitch lag (adaptive codebook index T) and pitch gain (adaptive codebook gain b). The adaptive codebook contribution is typically a past excitation at delay T or its interpolated version. The adaptive codebook index T is encoded and sent to the remote decoder. The pitch gain b is also quantized and sent to the remote decoder.

When closed-loop pitch analysis has been completed in operation 261 and fixed codebook contributions have been used, the calculator 105 for time-domain excitation contributions includes searching a fixed algebraic codebook 212 during operation 262 to find optimal fixed codebook parameters, typically including a fixed codebook index and a fixed codebook gain. The fixed codebook index and gain form the fixed codebook contribution. The fixed codebook index is encoded and sent to a remote decoder. The fixed codebook gain is also quantized and sent to a remote decoder. Fixed algebraic codebooks and their search are considered well known to those skilled in the art of CELP coding and will therefore not be described further in this disclosure.

Adaptive codebook index and gain, as well as fixed codebook index and gain (if used), form the temporal CELP excitation contribution.

4) Frequency conversion

During frequency domain encoding in a hybrid time/frequency domain decoding mode, two signals are represented in the transform domain (e.g., in the frequency domain). In one embodiment, a 256-point Type II (or Type IV) DCT (Discrete Cosine Transform) can be used to implement the time-to-frequency transformation (this DCT provides a resolution of 25 Hz with an internal sampling rate of 12.8 kHz), but any other suitable transform can be used. When using another transform, it may be necessary to modify the frequency resolution (defined above), the number of frequency bands, and the number of frequency segments per band (further defined below) accordingly.

As described above, in the unified time-domain/frequency-domain CELP encoding apparatus 100 and method 150 (Figures 1 and 2), a hybrid time-domain/frequency-domain encoding mode is used when the selector 205 selects general audio as the classification of the input sound signal 101 and no time attack is detected in the detector 208. Alternatively, in the unified time-domain/frequency-domain CELP encoding apparatus 700 and method 750 (Figure 7), a hybrid time-domain/frequency-domain encoding mode is used when the sound signal classifier 701 classifies the input sound signal 101 into the "unclear signal type" category. The hybrid time-domain/frequency-domain encoder 120/720 includes a calculator 107 (Figures 1 and 7) for frequency-domain excitation contributions, which performs an operation 157 to calculate the frequency-domain excitation contributions in response to the input LP residual res(n) (reference [5]) generated by the operation 251 of the LP analysis of the input sound signal 101 performed by the analyzer 201 (and the preprocessor 702). As shown in Figure 2, calculator 107 can calculate DCT 213, such as type IIDCT of the input LP residual res(n). The hybrid time-domain/frequency-domain encoder 120/720 also includes calculator 106 (Figures 1 and 7) for performing operation 156 to calculate the frequency transformation of the time-domain excitation contribution. As shown in Figure 2, calculator 106 can calculate DCT 214, such as type IIDCT of the time-domain excitation contribution. The frequency transformation of the input LP residual f _res and the time-domain CELP excitation contribution f _exc can be calculated using, for example, the following expression:

as well as:

Where r _{_es} (n) is the input LP residual, c _{_td} (n) is the temporal excitation contribution, and N is the frame length. In a possible implementation, for the corresponding internal sampling rate of 12.8 kHz, the frame length is 256 samples. The temporal excitation contribution is given by the following relationship:

e _{_td} (n) = bv(n) + gc(n)

Where v(n) is the adaptive codebook contribution, b is the adaptive codebook gain, c(n) is the fixed codebook contribution, and g is the fixed codebook gain. It should be noted that the temporal excitation contribution may consist solely of the adaptive codebook contribution as described above.

5) Cutoff frequency of time-domain contribution

When a sound signal sample is classified as general audio (Figure 1) or as an "unclear signal type" category (Figure 7), the contribution of time-domain excitation is not always significant for coding improvement compared to frequency-domain coding. Typically, it does improve coding in the lower part of the spectrum, while the improvement in coding in the higher part of the spectrum is minimal. The hybrid time-domain/frequency-domain encoder 120/720 includes a cutoff frequency lookup unit and filter 108 (Figures 1 and 7) for performing the operation 158 of determining a cutoff frequency, above which the coding improvement provided by the time-domain excitation contribution becomes too low to be valuable. As shown in Figure 2, the cutoff frequency lookup unit and filter 108 includes a cutoff frequency calculator 215 and a filter 216.

The operation 265, estimating the cutoff frequency of the time-domain excitation contribution, is first performed by calculator 215 (Fig. 2) using calculator 303 (Figs. 3 and 4). Calculator 303 performs a normalized cross-correlation operation 353 between the frequency transformation of the input LP residual 301 from calculator 107 and the frequency transformation of the time-domain excitation contribution 302 from calculator 106 (specified as _{f_res} and _{f_exc} , respectively, as defined in the preceding section 4). The last frequency _{L_f} in each of, for example, sixteen (16) frequency bands is defined in Hz as:

For this illustrative example, for a 20ms frame with an internal sampling rate of 12.8kHz, the number of frequency bins j per band B _b i, the cumulative frequency segments of band C _Bb , and the normalized cross-correlation C _c (i) of band i are defined, for example, as follows:

in

as well as

Where B _{_b} is the number of frequency segments j in each frequency band _{B b}, C _{_Bb} is the cumulative frequency segment in each frequency band, C _{_c} (i) is the normalized cross-correlation of each frequency band i, is the excitation energy of the frequency band, and similarly, is the residual energy of each frequency band.

The cutoff frequency calculator 215 includes a cross-correlation smoother 304 (Figures 3 and 4) that performs operations 354 to smooth the cross-correlation vectors between different frequency bands. More specifically, the cross-correlation smoother 304 uses, for example, the following relationship to calculate a new cross-correlation vector.

In the illustrative embodiments,

α=0.95; δ=(1-α); N _b =13; β=δ/2

The cutoff frequency calculator 215 also includes a calculator 305 (Figures 3 and 4) that performs the operation 355 of calculating the average value of the new cross-correlation vector over the first _Nb frequency bands (e.g., _Nb = 13, representing 5575 Hz).

The cutoff frequency calculator 215 also includes a cutoff frequency module 306 (FIG. 3), as shown in FIG. 4. The cutoff frequency module 306 includes a cross-correlation limiter 406, a cross-correlation normalizer 407, and a finder 408 for the frequency band with the lowest cross-correlation. More specifically, the limiter 406 performs an operation 456 limiting the average value of the cross-correlation vector to a minimum of 0.5, and the normalizer 407 performs an operation 457 normalizing the finite average value of the cross-correlation vector between 0 and 1. The finder 408 performs an operation 458 obtaining a first estimate of the cutoff frequency by finding the last frequency _Lf of frequency band i, which minimizes the difference between the last frequency _Lf of frequency band i and half the normalized average value of the cross-correlation vector multiplied by the internal sampling rate (Fs/2) of the input audio signal 101.

And among them

F ₅ = 12800Hz and

In the above relationship, represents the first estimate of the cutoff frequency.

At low bit rates, where the normalized average will never be very high (in the case of the unified time-domain/frequency-domain coding device 100 and method 150 of Figure 1), or when the submode flag _{F_tfsm} is greater than “0”, meaning the input audio signal is classified as an “unclear signal type” (in the case of the unified time-domain/frequency-domain coding device 700 and method 750 of Figure 7), or to artificially increase the value to contribute more weight to the time-domain excitation, the normalizer 407 can be used to amplify the value of the normalized average with a fixed scaling factor. As a non-limiting example, at bit rates below 8 kbps, the first estimate of the cutoff frequency is multiplied by 2.

The accuracy of the cutoff frequency can be improved by adding the following components to the calculation. For this purpose, the cutoff frequency module 306 includes an extrapolator 410 (Figure 4) in corresponding operation 460 that calculates the eighth harmonic using, for example, the minimum or lowest pitch lag value of the temporal excitation contribution from the subframe of the frame using the following relationship:

Where F _{_s} = 12800Hz is the internal sampling rate or frequency, N _{_sub} is the number of subframes in the frame, and T(i) is the adaptive codebook index or tone lag of subframe i.

The cutoff frequency module 306 includes a finder 409 (Figure 4) for the frequency band containing the eighth harmonic. More specifically, for subframe i < N _sub , the finder 409 performs a search operation 459 for the highest frequency band, for which, for example, the following inequality is still verified:

The index of this frequency band will be called and it indicates the frequency band where the 8th harmonic may be located.

The cutoff frequency module 306 ultimately includes a selector 411 for the final cutoff frequency f _{_tc} (Figure 4). More specifically, selector 411 performs an operation 461 using the following relationship to maintain the higher frequency between the first estimated cutoff frequency f _{_tc1} from finder 408 and the last frequency of the band containing the eighth harmonic from finder 409:

f _tc =max(L _f (i _8th ), f _tc1 )

When using a coding sub-mode, in the case of the unified time-domain/frequency-domain coding device 700 and method 750 of Figure 7, the cutoff frequency f _{_tc} is further thresholded using, for example, the following relationship:

f _tc =max(max(L _f (i _8th ),2775),f _tc1 )

As shown in Figures 3 and 4:

- The cutoff frequency calculator 215 also includes: a determiner 307 (Figure 3) for performing an operation 357 to determine the number of frequency bands to be zeroed;

- The decision maker 307 itself includes: an analyzer 415 (Figure 4) for performing parameter analysis operation 465; and a selector 416 (Figure 4) for performing selection of the frequency band to be zeroed operation 466; and

- Filter 216 (Figure 2) operates in the frequency domain and includes a zeroer 308 (Figure 3) for performing filtering operation 266. The corresponding operation 358 zeroes the frequency range determined to be zeroed in the decision 307. Zeroer 308 can zero (a) all frequency ranges (zeroer 417 and corresponding zeroing operation 467 in Figure 4) or (b) frequencies with a supplementary smooth transition region located at the cutoff frequency f_{_tc.}

The higher frequency range above (filter 418 and corresponding filter operation 468 in Figure 4). The transition region is located above the cutoff frequency _ftc and below the return-to-zero band, and it allows for a smooth spectral transition between the unchanged spectrum below the cutoff frequency _ftc and the return-to-zero band in the higher frequencies.

As a non-limiting illustrative example, when the cutoff frequency f _{_tc} from selector 411 is below or equal to 775Hz, analyzer 415 considers the cost of the time-domain excitation contribution too high. Selector 416 then selects all frequency segments representing the frequency of the time-domain excitation contribution to be zeroed, and zeroer 417 forces all frequency segments to zero, and also forces the cutoff frequency f _{_tc} to zero. All bits allocated to the time-domain excitation contribution are then reassigned to the frequency-domain coding pattern. Otherwise, analyzer 415 forces selector 416 to select higher frequency segments above the cutoff frequency f_{_tc} so that they can be zeroed by filter (zeroer) 418.

Finally, the cutoff frequency calculator 215 includes a quantizer 309 (Figures 3 and 4) for performing an operation 359 to quantize the cutoff frequency f _{_tc} to a quantized version f _{_tcQ} of that cutoff frequency for transmission to a remote decoder. For example, if three (3) bits are associated with the cutoff frequency parameter, a set of possible output values (in Hz) can be defined as follows:

f _tcQ ={0,1175,1575,1975,2375,2775,3175,3575}

Selector 411 can use a number of mechanisms to stabilize the selection of the final cutoff frequency f_{_tc} to prevent the quantized version f _{_tcQ} from switching between 0 and 1175 in an inappropriate signal segment. To achieve this, as a non-limiting example, analyzer 415 responds to the long-term average tone gain G _{_lt} 412 from the closed-loop tone analyzer 211 (Figure 2), the open-loop tone correlation C _{_ol} 413 from the open-loop tone analyzer 203, and the smoothed open-loop tone correlation C _{_st} 414. To prevent switching to frequency domain coding only, analyzer 415 disallows such frequency domain coding only if, for example, f_{_tcQ} cannot be set to 0:

f _tc > 2375Hz or

f _{_tc}> 1175Hz and C _{_ol}> 0.7 and G _{_lt}> 0.6

or

f _{_tc} ≥ 1175 Hz and C _{_st}> 0.8 and G _{_lt} ≥ 0.4

or

f _{_tcQ} (t-1)! = 0 and C _{_ol} > 0.5 and C _{_st} > 0.5 and G _{_lt} ≥ 0.6

Where c_{_ol} is the open-loop pitch correlation 413, and C _{_st} corresponds to a smoothed version of the open-loop pitch correlation 414 as defined by C _{_st} = 0.9·C _{_ol} + 0.1·C_{_st} . Furthermore, G _{_lt} (term 412 in Figure 4) corresponds to the long-term average of the pitch gain obtained by the closed-loop pitch analyzer 211 within the temporal excitation contribution. The long-term average of the pitch gain 412 is defined as where is the average pitch gain over the current frame. To further reduce the switching rate between frequency-only coding and hybrid temporal/frequency coding, a delay can be added.

6) Frequency domain coding

6.1) Create the difference vector

Once the cutoff frequency f _{_tc} of the time-domain excitation contribution is determined, frequency-domain coding is performed. To perform this frequency-domain coding, the hybrid time-domain/frequency-domain coding method 170/770 includes a subtraction operation 159, a frequency quantization operation 160, and an addition operation 161. The hybrid time-domain/frequency-domain encoder 120/720 includes a subtractor or calculator 109, a frequency quantizer 110, and an adder 111 to perform operations 159, 160, and 161, respectively.

Figure 5 is a schematic block diagram showing an overview of both the frequency quantizer 110 and the corresponding frequency quantization operation 160. Furthermore, Figure 6 is a schematic block diagram showing a more detailed structure of the frequency quantizer 110 and the corresponding frequency quantization operation 160.

Subtractor or calculator 109 (Figures 1, 2, 5, and 6) uses the difference between the frequency transform _fres 502 (Figures 5 and 6) (or other frequency representation) of the input LP residual from DCT 213 (Figure 2) and the frequency transform _fexc 501 (Figures 5 and 6) (or other frequency representation) of the time-domain excitation contribution from DCT 214 (Figure 2) to form the first part of the difference vector f, which ranges from zero to the cutoff frequency _ftc of the time-domain excitation contribution. Before the corresponding spectral portion of the frequency transform _fres 502 is subtracted from there, a reduction factor 603 (Figure 6) can be applied (see multiplier 604 and the corresponding multiplication operation 654) to the frequency transform _fexc 501 for the next transition region (80 frequency bands in this implementation example) where _ftrans = 2kHz. The result of the subtraction constitutes the second part of the difference vector _fd , which represents the frequency range from the _{cutoff frequency ftc to ftc} + _ftrans . The frequency transformation _{f_res502} of the input LP residual is used for the remaining third part of the difference vector _{f_d} .

The reduction portion of the difference vector f_{_d} generated by applying a reduction factor of 603 can be performed using any type of fade-out function. It can be shortened to only a few frequency bands, but it can also be omitted when the available bit budget is deemed sufficient to prevent energy oscillation artifacts when the cutoff frequency f_{_tc} changes. For example, at 25Hz resolution, corresponding to one frequency band f _{_bin} = 25Hz in a 256-point DCT at an internal sampling rate of 12.8kHz, the difference vector can be constructed as follows:

f _d (k) = f _res (k) - f _exc (k)

where 0≤k≤f _tc /f _bin

where f _tc /f _bin <k≤(f _tc +f _trans )/f _bin

f _d (k) = f _res (k), otherwise

Among them, f _res , f _exc and f _tc have been defined in the previous description.

6.2) Frequency domain bit allocation for coded sub-modes

6.2.1) Allocate a portion of the available bits to lower frequencies.

In the unified time-domain/frequency-domain CELP coding method 750 shown in Figure 7, the hybrid time-domain/frequency-domain encoder 720 includes a frequency band selector and a bit allocator 707, and the hybrid time-domain/frequency-domain coding method 770 includes corresponding operations of frequency band selection and bit allocation detection 757.

Figure 9 is a schematic block diagram showing the band selector and bit allocator 707 of Figure 7, as well as the corresponding operation 757 of band selection and bit allocation, which is used to allocate the available bit budget to the frequency quantization of the difference vector _fd when the input audio signal 101 is neither classified as speech nor as music in an alternative implementation of the unified time-domain/frequency-domain CELP coding method 150/750 of Figures 7 and 8.

Specifically, Figure 9 illustrates an innovative approach to how the band selector and bit allocator 707 can allocate available bits to frequency quantization when the input audio signal 101 is not classified as speech or music, but rather as an "unclear signal type" based on a previously selected coding sub-pattern. In Figure 9, frequency quantization is performed per band. For simplicity, in the current illustrative example, the bands have the same number of frequency segments, which is sixteen (16) frequency segments with an internal sampling rate of 12.8 kHz. Band "0" represents the lower portion of the spectrum, while band "15" represents the higher portion of that spectrum.

To best utilize the bits available for frequency quantization, the band selection and bit allocation operation 757 includes a first operation 951, which pre-fixes a portion of the available bit budget (see 900) for quantizing the lower frequency of the difference vector _fd as a function of the quantized cutoff frequency f _{_tcQ} from the cutoff frequency finder and filter 108. To perform operation 951, the estimator 901 uses, for example, the following relationship:

P _Btf =max(min(P _Btf ,0.75),0.5)

Where P _{_Blf} is a portion of the available bits for frequency quantization allocated to the lower frequency of the difference vector f_{_d}. In this example, the lower frequency refers to the first five (5) frequency bands, or the first two (2) kHz. The term L _{_f} (f_{_tcQ} ) refers to the number of frequency bands up to the quantization cutoff frequency f _{_tcQ} .

Then, estimator 901 adjusts the portion of available bits P _{_Blf} allocated to the lower frequency frequency quantization based on the coded sub-mode flag F _{_tfsm}. If the coded sub-mode flag F _{_tfsm} is set to "2" (Figure 8), indicating a possibility of a time attack being detected in the current frame of the input audio signal 101, the portion of available bits allocated to the lower frequency frequency quantization increases by 10%. If a "musical" feature is detected in the content of the current frame, as indicated by the sub-mode coding flag F _{_tfsm} set to "3", the portion of available bits P _{_Blf} allocated to the lower frequency frequency quantization decreases by 10%.

10% of the special amount.

6.2.2) Estimate the number of frequency bands to be quantized

Another parameter affecting the total number of bits per band that can be used for frequency quantization of the difference vector f _d is the maximum number of estimated bands N _Bmx for the difference vector f _d to be quantized. In the illustrative example described here, at an internal sampling rate of 12.8 kHz, the maximum total number of bands N _tt is sixteen (16).

When using a coding submode, the band selection and bit allocation operation 757 includes an operation 952 that estimates the maximum number of bands N _Bmx for the difference vector f _d to be quantized. To perform operation 952, if the coding submode flag F _tfsm is set to "1" (selecting the first coding submode), the estimator 902 sets the maximum number of bands N _Bmx to "10". If the coding submode flag F _tfsm is set to "2" (selecting the second coding submode), then the estimator 902 sets the maximum number of bands N _Bmx to "9". If the coding submode flag F _tfsm is set to "3" (selecting the third coding submode), then the estimator 902 sets the maximum number of bands N _Bmx to "13". The estimator 902 then readjusts the maximum number of bands N _Bmx to be quantized using, for example, the following relationship as a function of the bit budget available for frequency quantization of the difference vector f _d :

N _Bmx =max(min(trunc(N _Bmx ·N _Badj +0.5), N _tt ), 5),

Where _BF represents the number of bits available for frequency quantization of the difference vector _fd (see 900), _BT is the total bit rate available for encoding the channel being processed (see 900), _Ftfsm is the submode flag (see 900), and _Ntt is the maximum total number of frequency bands.

Estimator 902 can further reduce the maximum number of frequency bands to be quantized for the difference vector f _d in relation to the number of quantization bits allocated to the mid-frequency and higher-frequency bands of the difference vector f _d . For this limitation, it is assumed that the last lower-frequency band and the first band thereafter have similar bit numbers m _b , or approximately 17% of the number of bits P _Blf allocated to the frequency quantization of the lower frequencies. For the last frequency band to be quantized, a minimum number m _p of 4.5 bits is used to quantize at least one (1) frequency pulse. If the available bit rate _BT is greater than or equal to 15 kbps, the minimum number of bits m _p will be nine (9) to allow more pulses to be quantized per frequency band. However, if the total available bit rate BT is less than 15 kbps, but the submode flag F _tfsm is set to “3”, meaning the content is similar to music, the number of bits m for the last frequency band to be quantized will be 6.75 to allow for more precise quantization. Estimator 902 then uses, for example, the following relationship to calculate the maximum number of corrected frequency bands N′ _Bmx :

N′ _Bmx =min(N _Bmx,5 +(B _F -P _Blf ·B _F )/0.5.(m _p +m _b ))

Where _N′Bmx corresponds to the maximum number of corrections for the frequency band to be quantized, _NBmx is the maximum number of estimated frequency bands, the number “5” represents the minimum number of frequency bands, _BF represents the number of bits available for frequency quantization of the difference vector _fd , _PBlf is a portion of the bits allocated to the quantization of the five (5) lower frequency bands, _mp is the minimum number of bits allocated to the frequency quantization of the frequency band, and _mb is the number of bits allocated to the quantization of the first frequency band after the five (5) lower frequency bands.

After calculating the maximum number of frequency bands, the estimator 902 can perform additional verification to ensure that _mp remains less than or equal to _mb . While this additional verification is an optional step, it helps to allocate bits more efficiently among the frequency bands of the difference vector _fd at low bit rates.

6.2.3) Modify the number of bits allocated to lower frequencies

The band selection and bit allocation operation 757 includes an operation 953 for calculating low-frequency bits. To perform operation 953, a calculator 903 is provided. If the calculation of the maximum number of bands N′ _Bmx results in a smaller number of bands to be quantized, the calculator 903 reallocates the bit portions previously allocated to higher frequency bands using, for example, the following relationship, such that they are no longer relevant to the quantization of lower frequency bands:

B _LF =P _Blf ·B _F +(0.5·(m _p +m _b )·(N _Bmx -N′ _Bmx )),

Where _BLF corresponds to the bits allocated to five (5) lower frequency bands, _BF corresponds to the number of bits available for frequency quantization of the lower frequency of the difference vector _fd , _PBlf is a portion of the aforementioned bits allocated to, for example, five (5) lower frequency bands for frequency quantization from estimator 901, _mp is the minimum number of bits allocated to the quantization bands, and _mb is the number of bits allocated to the first band after quantizing five (5) lower frequency bands.

6.2.4) Dual sorting of frequency bands

The band selection and bit allocation operation 757 includes a band characterization operation 954. To perform operation 954, the band selector and bit allocator 707 includes a band characterizer 904 that, once the bit rate is allocated between lower bands and the remaining bands, performs a dual sorting of the bands to determine the importance of each band. The first sorting involves finding one or more bands that have lower energy compared to their neighboring bands. When this occurs, the characterizer 904 marks these bands such that even if the available bit budget is high, only a predetermined minimum number of bits _mp can be allocated to these low-energy bands for frequency quantization. The second sorting involves performing a positional sorting of the medium-energy and higher-energy bands, for example, in descending order of energy. These first and second sortings (dual sorting) are not performed on the lower bands, but rather up to the maximum number of bands _N′Bmx . The band characterization operation 954 can be summarized as follows:

Where P _{_pb} (i) is set to "1" for the frequency band that will use only the minimum number of bits m_{_p}, including the positions of the medium-energy and high-energy frequency bands in descending order of energy, and E(i) corresponds to the energy of each frequency band. C _{_Bb} and B_{_b} are defined in Section 5 above. The difference vector f_{_d} has been defined in Section 6.1.

In calculator 708 and corresponding operation 758 of Figures 7 and 9, the energy E(i) of each frequency band of the difference vector f _d is calculated. Calculator 708 and operation 758 also calculate the gain per frequency band, as described with reference to calculator 615 and operation 665 of Figure 6. The energy E(i) of each frequency band of the difference vector f _d and the gain of each frequency band are quantized, for example, as described with reference to quantizer 616 and operation 666 of Figure 6, and both are sent to a remote decoder. In the implementation of Figure 7 for the unified time-domain/frequency-domain coding device 700 and method 750, calculator 708 and operation 758 replace calculator 615 and operation 665 and quantizer 616 and operation 666.

6.2.5) Allocate bits to the selected frequency band

The band selection and bit allocation operation 757 includes a final bit allocation operation 955 per band. To perform operation 955, the band selector and bit allocator 707 includes a final bit allocator 905 per band.

Once the frequency bands have been characterized, the allocator 905 allocates the bit rate or number of bits _BF that can be used to perform frequency quantization of the difference vector _fd between the selected frequency bands.

In a non-limiting example, for the first five (5) lower frequency bands, the allocator 905 linearly allocates the bits B _LF assigned for frequency quantization of the lower frequencies, wherein the first lowest frequency band receives 23% of the bits B _LF and the fifth (5th) lower frequency band receives the last 17% of the bits B _LF . In this way, the lower frequencies of the spectrum of the difference vector f _d can be quantized with sufficient precision to recover a better quality synthesis of the input audio signal 101.

Allocator 905 distributes the remaining bits _BF allocated for frequency quantization of the difference vector f _{_d} according to a linear function across another, mid-frequency band, and higher-frequency bands. However, again considering the previous band energy characterization (operation 954), more bits can be allocated to higher-energy bands and fewer bits to lower-energy bands compared to their adjacent bands, thus making more relevant use of the available bits by quantizing the more important parts of the spectrum of the difference vector f _{_d} with higher precision. As a non-limiting example, the following relationship illustrates how bit allocation (operation 955) can be performed:

Where _Bp (i) represents the number of bits allocated to each frequency band i, _BF represents the number of bits available for the frequency quantization difference vector _fd , _BLF corresponds to the bit rate or bits allocated to the five (5) lower frequency bands, _mp is the minimum number of bits for the frequency pulse in the quantization band, _Ppb (i) contains the position where the minimum number of bits _mp will be used, and _N′Bmx is the maximum number of frequency bands to be quantized.

If there are some unallocated bits after operation 955, allocator 905 allocates them to lower frequency bands. As a non-limiting example, allocator 905 will allocate one remaining bit per band, starting from the fifth (5th) frequency band and going back to the first frequency band, and repeat the process if all remaining bits need to be allocated.

Later, the allocator 905 may have to round, truncate, or round the number of bits per frequency band based on the algorithm used to perform the quantization of the frequency pulses and the potential fixed-point implementation.

6.3) Search frequency pulse

The hybrid time-domain/frequency-domain CELP coding method 170/770 includes the operation of frequency quantization 160 (Figures 1, 2, and 7) of the difference vector f _d . In order to perform operation 160, the hybrid time-domain/frequency-domain CELP encoder 120/720 includes a frequency quantizer 110 (219 in Figure 2).

Several methods can be used to quantize the difference vector f_{_d} . In each case, frequency pulses must be searched and quantized. In one possible implementation, the frequency quantizer 110 searches across the spectrum for the highest energy pulse of the difference vector f_{_d}. The method for searching for pulses can be as simple as dividing the spectrum into bands and allowing a certain number of pulses in each band. The number of pulses in each band depends on the available bit budget and the position of the band within the spectrum. Typically, more pulses are allocated to lower frequencies.

6.4) Quantized difference vector

Depending on the available bit rate, the quantization of the frequency pulses can be performed by the frequency quantizer 110 using different techniques. In one embodiment, at bit rates below 12 kbps, a simple search and quantization scheme can be used to encode the position and symbol of the pulse. This scheme is described below as a non-limiting example.

For frequencies below 3175 Hz, a simple search and quantization scheme is used using a factorial pulse coding (FPC) method, which is described in the literature, for example in reference [8], the entire contents of which are incorporated herein by reference.

More specifically, referring to Figures 5 and 6, the frequency quantizer 110 includes a selector 504 to perform an operation 554 of determining whether to use FPC quantization for all the spectrum. As shown in Figure 5, if the selector 504 determines that FPC quantization for all the spectrum is not used, then FPC encoding and pulse position and symbol encoding operations 556 are performed in the encoder 506.

As shown in Figure 6, the FPC encoding and pulse position and sign encoding operation 556 includes a frequency pulse search operation 659, an FPC encoding operation 660, an operation to find the highest energy pulse 661, and an operation to quantize the position and sign of the frequency pulse 662. To perform operations 659-662, the encoder 506 includes a frequency pulse searcher 609, an FPC encoder 610, a highest energy pulse finder 611, and a frequency pulse position and sign quantizer 612, respectively.

Searcher 609 searches for frequency pulses below 3175 Hz across all frequency bands. FPC encoder 610 then processes the frequency pulses. Searcher 611 identifies the highest energy pulse for frequencies equal to and greater than 3175 Hz, and quantizer 612 encodes the position and sign of the found highest energy pulse. If more than one (1) pulse is allowed within the frequency band, the amplitude of the previously found pulse is divided by 2, and the search is repeated across the entire frequency band. Each time a pulse is found, its position and sign are stored for use in the quantization and bit packing stages. As a non-limiting example, the following pseudocode illustrates this simple search and quantization scheme:

Where _NBD is the number of frequency bands ( _NBD = 16 in the illustrative example), _Np is the number of pulses i to be encoded in frequency band k, _Bb is the number of frequency segments in each frequency band, _CBb is the cumulative frequency segment of each frequency band as defined previously in Section 5, _pp represents the vector containing the location of the found pulse, _ps represents the vector containing the symbol of the found pulse, and _pmax represents the energy of the found pulse.

At bit rates above 12 kbps, selector 504 determines which spectrum to quantize using FPC (Figures 5 and 6). As shown in Figure 5, FPC encoding operation 555 is then performed in FPC encoder 505. Referring to Figure 6, encoder 505 includes frequency pulse searcher 607, and operation 555 includes a corresponding operation 667 for searching for frequency pulses. The search for frequency pulses is performed across the entire frequency band. Operation 555 includes operation 668 for encoding the found frequency pulses, and encoder 505 includes FPC processor 608 for performing operation 668.

Then, the FPC processor 608 or the quantizer 612 for pulse positions and symbols obtains the quantized difference vector f _{_dQ} by adding the number of pulses n_{b_pulses} with pulse symbols p_{_s} to each found position p_{_p} . For each frequency band, the quantized difference vector f _{_dQ} can be written using, for example, the following pseudocode:

for j = 0, ..., j < nb_pulses

f _dQ (p _p (j))+＝p _s (j)

6.5) Noise Filling

The frequency band is quantized with more or less precision; the quantization method described in the previous section does not guarantee that all frequency segments within the band are quantized. This is especially true at low bit rates where the number of pulses quantized in each band is relatively low. To prevent audible artifacts caused by these unquantized frequency segments, the frequency quantizer 110 includes a noise filler 507 (FIG. 5) to perform a corresponding operation 557 of adding some noise to the unquantized frequency segments to fill these gaps. For example, this noise addition can be performed across the entire spectrum at bit rates below 12 kbps, but at higher bit rates, it can be applied only above the cutoff frequency f _{_tc} contributed by the time-domain excitation. For simplicity, the noise intensity varies only with the available bit rate. At high bit rates, the noise level is low, but at low bit rates, the noise level is high.

The noise filler 507 includes an adder 613 (FIG. 6) that performs an operation 663 of adding noise to the quantized difference vector f _dQ after the strength or energy level of the added noise has been determined. For this purpose, the frequency quantization operation 160 includes an operation 664 of estimating the strength or energy level of the added noise, and to perform operation 664, the frequency quantizer 110 includes a corresponding estimator 614 for the noise energy level. The operation 664 of estimating the strength or energy level of the added noise is performed by the estimator 614 and precedes the operation 665 of determining the per-band gain in the per-band gain calculator 615 of the frequency quantizer 110.

In the illustrative embodiment, in estimator 614, the noise level is directly related to the coding bit rate. For example, at 6.60 kbps, estimator 614 sets the noise level _N′L to 0.4 times the amplitude of the frequency pulse encoded in a specific frequency band, and gradually decreases it to a value of 0.2 times the amplitude of the frequency pulse encoded in a 24 kbps frequency band. Adder 613 injects noise only into portions of the spectrum where a certain number of consecutive frequency bands have very low energy, for example, when the cumulative energy of half of the frequency band is below 0.5. For a specific frequency band i, noise is injected, for example, as follows:

Where, for frequency band i, C _{_Bb} is the cumulative number of frequency segments in each frequency band, B_{_b} is the number of frequency segments in a specific frequency band i, _N′L is the level of added noise, and r _and are random number generators restricted to between -1 and 1.

6.6) Per-band gain quantization

Referring to Figures 5 and 6, the frequency quantization operation 160 of the unified time-domain/frequency-domain coding device 100 and method 150 includes an operation 665 for determining the gain per frequency band, followed by an operation 666 for quantizing the gain per frequency band. To perform operations 665 and 666, the frequency quantizer 110 includes a gain per frequency band calculator 615 and a gain per frequency band quantizer 616.

Once the quantized difference vector f _dQ is found (including noise fill if necessary), calculator 615 calculates the per-band gain for each frequency band. The per-band gain for a specific frequency band _Gb (l) is defined as the ratio between the energy of the unquantized difference vector f _dQ and the energy of the quantized difference vector f _dQ in the logarithmic domain, using, for example, the following relationship:

C _{_Bb} and B_{_b} are defined in Part 5 above.

The per-band gain quantizer 616 vector-quantizes the per-band frequency gain. Prior to vector quantization, at low bit rates, the last gain (corresponding to the last band) is quantized individually, and the remaining fifteen (15) per-band gains (e.g., when using 16 bands) are divided by the quantized last gain. The quantizer 616 then performs vector quantization on the normalized fifteen (15) remaining gains. At higher bit rates, the average per-band gain is first quantized, and then that average is removed from all per-band gains across, for example, sixteen (16) bands before vector quantization of these per-band gains. The vector quantization used can be a standard minimization in the logarithmic domain of the distance between the vector containing the per-band gain and entries in a particular codebook.

In the frequency domain coding mode, a gain is calculated for each frequency band in calculator 615 to match the energy of the unquantized vector f_{_d} with the quantized vector f_{_d} Q. The gain is vector-quantized in quantizer 616 and applied to the quantized vector f _{_dQ} in each frequency band (operation 559) via multiplier 509 (Figures 5 and 6).

Alternatively, the entire spectrum can be encoded using an FPC scheme at a rate below 12 kbps by selecting only a subset of the frequency bands to be quantized. Before selecting the frequency bands, the energy _Ed of the unquantized difference vector _fd is quantized using quantizer 616. The energy is calculated using, for example, the following relationship:

E _{_d} (i) = log _₁₀(S _{_d} (i))

C _{_Bb} and B_{_b} are defined in Part 5 above.

To perform quantization of the band energy _Ed ′, the average energy over the first 12 of the sixteen bands used is first quantized, and the average energy over the first 12 of the sixteen bands used is subtracted from the total sixteen (16) band energy. Then, all bands are vectors quantized in groups of 3 or 4 bands. The vector quantization used can be a standard minimization in the logarithmic domain of the distance between the vector containing the gain of each band and the entries of a particular codebook. If not enough bits are available, only the first 12 bands can be quantized, and the average of the first three (3) bands can be used, or the last four (4) bands can be extrapolated by any other method.

Once the energy of the unquantized difference vector's frequency bands has been quantized, the energy can be sorted in descending order, making it reproducible on the decoder side. During sorting, all energy bands below 2kHz are always kept, and only the highest energy bands are then passed to the FPC scheme for encoding frequency pulse amplitudes and symbols. Using this method, the FPC scheme encodes smaller vectors but covers a wider frequency range. In other words, it requires fewer bits to cover important energy events across the entire spectrum.

In the specific implementation of the unified time-domain/frequency-domain coding device 700 and method 750 of FIG7, band selection and bit allocation are performed alternatively as determined by the per-band energy and per-band gain calculator 708 and calculation operation 758 of FIG7 and 9 as described above, and the band selector and bit allocator 707 and band selection and bit allocation operation 757.

After pulse quantization, noise filling, similar to that described previously, is performed. Then, a gain adjustment factor _Ga is calculated for each frequency band to match the energy _EdQ of the quantized difference vector f _dQ to the quantized energy Ed _′ of the unquantized difference vector f _d . This per-band gain adjustment factor is then applied to the quantized difference vector f _dQ . This can be represented as follows:

in,

as well as

E′ _d is the quantized per-band energy of the unquantized difference vector f _d as defined earlier.

After the frequency domain coding phase is completed, the total time-domain/frequency-domain excitation is found. For this purpose, the hybrid time-domain/frequency-domain CELP coding method 170/770 includes an operation 161 in which the difference vector f _{_dQ} from the frequency quantizer 110 is added to the filtered frequency-transformed time-domain excitation contribution f _{_excF} using adder 111 (Figures 1, 2, 5, and 6) of the hybrid time-domain/frequency-domain CELP encoder 120/720. When the unified time-domain/frequency-domain coding device 100/700 changes its bit allocation from a time-domain-only coding mode to a hybrid time-domain/frequency-domain coding mode, the per-band excitation spectrum energy of the time-domain-only coding mode does not match the per-band excitation spectrum energy of the hybrid time-domain/frequency-domain coding mode. This energy mismatch can produce switching artifacts that are more audible at low bit rates. To reduce any audible degradation caused by this bit reallocation, a long-term gain can be calculated for each band, and the long-term gain can be applied to the summed excitation to correct the energy of each band for several frames after the reallocation. Then, the hybrid time-domain/frequency-domain CELP coding method 170/770 includes operations 162 (Figures 1, 5 and 6) for transforming the sum of the frequency-quantized difference vector f _dQ and the frequency-transformed and filtered time-domain excitation contribution f _excF to the time domain using, for example, IDCT (inverse DCT) 220 (Figure 2).

The unified time-domain/frequency-domain coding method 150/750 includes the operation 163/756 of filtering the total time-domain/frequency-domain excitation from the IDCT 220 through the LP synthesis filter 113/706 (Figures 1, 2 and 7) of the coding device 100/700 to generate a synthesized signal.

The quantized position and sign of the frequency pulse that forms the quantized difference vector f _dQ are sent to a remote decoder (not shown).

In one non-limiting embodiment, while the CELP-coded memory is updated on a subframe-by-subframe basis using only temporal excitation contributions, total temporal/frequency excitation is used to update those memories at frame boundaries. In another possible implementation, the CELP-coded memory is updated on a subframe-by-subframe basis and also only using temporal excitation contributions at frame boundaries. This results in an embedded architecture where the frequency-quantized signal constitutes an upquantization layer independent of the core CELP layer. This presents advantages in certain applications. In this particular case, a fixed codebook is always used to maintain good perceptual quality, and the number of subframes is always four (4) for the same reason. However, frequency domain analysis can be applied to the entire frame. This embedded approach is suitable for approximately 12 kbps and higher bit rates.

7) Decoder devices and methods

Figure 11 is a schematic block diagram showing a decoder device 1100 and a corresponding decoding method 1150 for decoding the bitstream 1101 from the aforementioned unified time-domain/frequency-domain coding device 700 and the corresponding unified time-domain/frequency-domain coding method 750.

The decoder device 1100 includes a receiver (not shown) for receiving bit stream 1101 from the unified time-domain/frequency-domain coding device 700.

If the audio signal encoded by the unified time-domain/frequency-domain encoding device 700 has been classified as "music", this is indicated by the corresponding signaling bits in the bitstream 1101 and detected by the decoder device 1100 (see 1102). The received bitstream 1101 is then decoded by the "music" decoder 1103 (e.g., a frequency-domain decoder).

If the audio signal encoded by the unified time-domain/frequency-domain coding device 700 has been classified as “speech,” this is indicated by the corresponding signaling bits in the bitstream 1101 and detected by the decoder device 1100 (see 1104). The received bitstream 1101 is then decoded by the “speech” decoder 1105, for example, using a time-domain decoder employing ACELP (Algebraic Code-Excited Linear Prediction) or, more generally, CELP (Code-Excited Linear Prediction).

If the audio signal encoded by the unified time-domain/frequency-domain coding device 700 has not yet been classified as "music" or "speech" (see 1102 and 1104), and the bit rate available for encoding the audio signal is equal to or less than 9.2 kbps (see 1106), this is indicated in the bitstream by a submode flag F _tfsm set to "0". The received bitstream 1101 is then decoded using a backward coding mode (i.e., the conventional unified time-domain and frequency-domain coding model (EVS) of Figures 1 and 2 as shown in 1107).

Finally, if the audio signal encoded by the unified time-domain/frequency-domain coding device 700 has not yet been classified as "music" or "speech" (see 1102 and 1104), and the bit rate available for encoding the audio signal is higher than 9.2 kbps (see 1106), this is indicated in bitstream 1101 by a sub-mode flag F _tfsm set to "1", "2", or "3". The received bitstream 1101 is then decoded using the audio signal decoder 1200 of FIG. 12 and the corresponding audio signal decoding method 1250.

7.1) Audio signal decoder and decoding method

Figure 12 is a schematic block diagram showing both the audio signal decoder 1200 and the corresponding audio signal decoding method 1250, which are used to decode bitstreams from the aforementioned unified time-domain/frequency-domain coding device 700 and the corresponding unified time-domain/frequency-domain coding method 750 when the audio signal is classified as having an unclear signal type category.

As mentioned in the preceding description, the adaptive codebook index T and the adaptive codebook gain b are quantized and transmitted, and thus received in the bitstream by a receiver (not shown). Similarly, when used, the fixed codebook index and fixed codebook gain are also quantized and transmitted to the decoder, and thus received in bitstream 1101 by a receiver (not shown). The audio signal decoding method 1250 includes an operation 1256 to calculate the temporal excitation contribution of the decoded signal using the adaptive codebook index and gain, as well as the fixed codebook index and gain (if used), as is typically done in the CELP coding field. To perform operation 1256, the audio signal decoder 1200 includes a calculator 126 for the temporal excitation contribution of the decoded signal.

The audio signal decoding method 1250 also includes an operation 1257 for calculating the frequency transformation of the decoded time-domain excitation contribution using the same procedure as in operation 156 using DCT transform. To perform operation 1257, the audio signal decoder 1200 includes a calculator 1207 for the frequency transformation of the decoded time-domain excitation contribution.

As mentioned in the preceding description, the quantized version of the cutoff frequency, f _{_tcQ} , is sent to the decoder and thus received by the receiver (not shown) in bitstream 1101. The audio signal decoding method 1250 includes an operation 1258 that filters the frequency transformation of the time-domain excitation contribution from the calculator 1207 using the decoded cutoff frequency f _{_tcQ} recovered from bitstream 1101 and a process identical or similar to the previously described filtering operation 266. To accomplish operation 1258, the audio signal decoder 1200 includes a filter 1208 that performs a frequency transformation of the time-domain excitation contribution using the recovered cutoff frequency f _{_tcQ}. Filter 1208 has the same or at least similar structure to filter 216 of FIG. 2.

The filtered frequency transformation of the time-domain excitation contribution from filter 1208 is provided to the positive input of adder 1209, which performs the corresponding addition operation 1259.

The audio signal decoding method 1250 includes an operation 1260 of calculating the decoded energy and gain for each frequency band of the difference vector f_{_d}. To perform operation 1260, the audio signal decoder 1200 includes a calculator 1210. Specifically, the calculator 1210 dequantizes the quantized energy and quantized gain for each frequency band received by a receiver (not shown) from a unified time-domain/frequency-domain coding device 700 in bitstream 1101 using a process opposite to that described in this disclosure.

Audio signal decoding method 1250 includes operation 1261 to recover the frequency-quantized difference vector f _dQ. To perform operation 1261, audio signal decoder 1200 includes calculator 1211. Calculator 1211 extracts the quantized position and symbol of frequency pulses from bitstream 1101 and copies the selection of frequency bands for quantization and bit allocation in different frequency bands determined by operation 757 and allocator 707 and used by unified time-domain/frequency-domain coding device 700 for encoding the input audio signal. Calculator 1211 uses this copied information to recover the frequency-quantized difference vector f _dQ from the extracted quantized position and symbol of the frequency pulses. Specifically, for this purpose, audio signal decoder 1200, in response to the number of bits (bit rate) available for frequency quantization of the difference vector f _dQ in decoder 1200 (see 1220), the total bit rate available for the channel being processed (see 1220), and the sub-mode flag (see 1220), copies the process used in unified time-domain/frequency-domain coding device 700 as shown in FIG. 9.

Specifically:

- Estimator 1201 and operation 1251 in Figure 12 correspond to estimator 901 and operation 951 in Figure 9, for pre-fixing a portion of the available bit budget for the lower frequency of the quantization difference vector _fd as a function of the quantized cutoff frequency f _tcQ .

- Estimator 1202 and operation 1252 in Figure 12 correspond to estimator 902 and operation 952 in Figure 9, and are used to estimate the maximum number of frequency bands N _Bmx of the quantized difference vector f _dQ .

- Calculator 1203 and operation 1253 in Figure 12 correspond to calculator 903 and operation 953 in Figure 9, and are used to calculate lower frequency bits.

Characterizer 1204 and operation 1254 in Figure 12 correspond to characterizer 904 and operation 954 in Figure 9 and are used for frequency band characterization.

The allocator 1205 and operation 1255 in Figure 12 correspond to the allocator 905 and operation 955 in Figure 9 for the final allocation of bits per frequency band.

The audio signal decoding method 1250 includes an operation 1259 of adding the difference vector f _dQ of the recovered frequency quantization from the calculator 1211 to the temporal excitation contribution f _excF of the frequency transformation and filtering from the filter 1208 to form a hybrid time-domain/frequency-domain excitation.

It is understood that estimators 1201 and 1202, calculator 1203, characterizer 1204, distributor 1205, calculators 1206 and 1207, filter 1208, calculators 1210 and 1211, and adder 1212 use information transmitted in bitstream 1101 to form a reconstructor with mixed time-domain/frequency-domain excitation, which includes a sub-mode flag identifying one of the coded sub-modes selected and used to encode sound signals classified as having an unclear signal type category.

In the same manner, operations 1251-1261 form a method for reconstructing the hybrid time-domain/frequency-domain excitation using the information transmitted in bitstream 1101.

The audio signal decoder 1200 includes a converter 1212 for performing an operation 1262 that transforms a mixed time-domain/frequency-domain excitation back to the time domain using, for example, an IDCT (inverse DCT) 220.

Finally, the synthesized audio signal is calculated in the decoder 1200 through the operation 1263 of filtering the total excitation from the converter 1212 by the LP (linear prediction) synthesis filter 1213. Of course, the LP parameters required by the decoder 1200 to reconstruct the synthesis filter 1213 are sent from the unified time-domain/frequency-domain coding device 700 and extracted from the bitstream 1101, as is well known in the field of CELP coding.

8) Hardware Implementation

Figure 10 is a simplified block diagram of an example configuration of the hardware components forming the above-described unified time-domain/frequency-domain coding device 100/700 and method 150/750, decoder device 1100 and decoding method 1150.

The unified time-domain/frequency-domain coding device 100/700 and decoder device 1100 can be implemented as part of a mobile terminal, a portable media player, or any similar device. Device 100/700 and decoder device 1100 (identified as 1000 in Figure 10) include an input 1002, an output 1003, a processor 1001, and a memory 1004.

Input 1002 is configured to receive the input audio signal 101/bit stream 1101 of Figures 1 and 7 in digital or analog form. Output 1003 is configured to provide an output signal. Input 1002 and output 1003 can be implemented in a common module, such as a serial input/output device.

Processor 1001 is operatively connected to input 1002, output 1003, and memory 1004. Processor 1001 is implemented as one or more processors for executing code instructions that support the functionality of various components of the unified time-domain/frequency-domain coding device 100/700 for encoding input audio signals as shown in Figures 1-9 or the decoder device 1100 of Figures 11-12.

Memory 1004 may include non-transitory memory for storing code instructions executable by processor 1001. Specifically, it includes/stores processor-readable memory for non-transitory instructions that, when executed, cause the processor to implement the operations and components of the unified time-domain/frequency-domain coding apparatus 100/700 and method 150/750, as well as the decoder apparatus 1100 and decoding method 1150 described in this disclosure. Memory 1004 may also include random access memory or buffers for storing intermediate processing data from various functions performed by processor 1001.

Those skilled in the art will recognize that the description of the unified time-domain/frequency-domain coding apparatus 100/700 and method 150/750, as well as the decoder apparatus 1100 and decoding method 1150, is merely illustrative and not intended to be limiting in any way. Other embodiments will readily conceive of those skilled in the art upon receiving this disclosure. Furthermore, the disclosed unified time-domain/frequency-domain coding apparatus 100/700 and method 150/750, decoder apparatus 1100, and decoding method 1150 can be customized to provide valuable solutions to existing needs and problems in encoding and decoding sound.

For clarity, not all routine features of the unified time-domain/frequency-domain coding devices 100/700 and 150/750, as well as the decoder device 1100 and decoding method 1150, are shown or described. It should be understood that in developing any such practical implementation of the unified time-domain/frequency-domain coding devices 100/700 and 150/750, as well as the decoder device 1100 and decoding method 1150, many implementation-specific decisions may need to be made to achieve the developer's specific goals, such as compliance with application, system, network, and business-related constraints, and these specific goals will vary from implementation to implementation and from developer to developer. Furthermore, it should be understood that development work may be complex and time-consuming, but it remains a routine engineering task for those skilled in the art of sound processing who benefit from this disclosure.

According to this disclosure, the components/processors/modules, processing operations, and/or data structures described herein can be implemented using various types of operating systems, computing platforms, network devices, computer programs, and/or general-purpose machines. Furthermore, those skilled in the art will recognize that less general-purpose devices, such as hardwired devices, field-programmable gate arrays (FPGAs), application-specific integrated circuits (ASICs), etc., can also be used. Where a method comprising a series of operations and sub-operations is implemented by a processor, computer, or machine, and those operations and sub-operations can be stored as a series of non-transitory code instructions readable by the processor, computer, or machine, they can be stored on tangible and/or non-transitory media.

The unified time-domain/frequency-domain coding apparatus 100/700 and method 150/750, as well as the decoder apparatus 1100 and decoding method 1150 described herein, may use software, firmware, hardware, or any combination of software, firmware, or hardware suitable for the purposes described herein.

In the unified time-domain/frequency-domain coding apparatus 100/700 and method 150/750, as well as the decoder apparatus 1100 and decoding method 1150 described herein, various operations and sub-operations can be performed in various orders, and some operations and sub-operations can be optional.

Although the present disclosure has been described above by way of its non-limiting illustrative embodiments, these embodiments may be modified freely within the scope of the appended claims without departing from the spirit and essence of the present disclosure.

9) References

This disclosure references the following sources, the entire contents of which are incorporated herein by reference:

[1] U.S. Patent 9,015,038, “Coding generic audio signals at low bit rate and low delay”.

[2] 3GPP TS26.445, v.12.0.0, “Codec for Enhanced Voice Services (EVS); Detailed Algorithmic Description”, September 2014.

[3] 3GPP SA4 Contribution S4-170749 “New WID on EVS Codec Extension for Immersive Voice and Audio Services”, SA4 Meeting #94, June 26-30, 2017, http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_94/Docs/S4-170749.zip

[4] U.S. Provisional Application 63/010,798, “Method and device for speech/music classification and core encoder selection in a sound codec”.

[5]ITU-T Recommendation G.718 “Frame error robust narrow-band and wideband embedded variable bit-rate coding of speech and audio from 8-32kbit/s”, June 2008.

[6] T. Vaillancourt et al., “Inter-tone noise reduction in a low bit rate CELP decoder”, Proceedings of the IEEE Proceedings of the International Conference on Acoustics, Speech and Signal Processing (ICASSP), Taipei, Taiwan, April 2009, pp. 4113-16.

[7] V. Eksler and M. Jelnek, “Transition mode coding for source controlled CELP codecs”, Proceedings of the IEEE Proceedings of the International Conference on Acoustics, Speech and Signal Processing (ICASSP), March-April 2008, pp. 4001-4043.

[8] U. Mittal, J.P. Ashley and E.M. Cruz-Zeno, “Low Complexity Factorial Pulse Coding of MDCT Coefficients using Approximation of Combinatorial Functions”, Proceedings of the IEEE Proceedings of the International Conference on Acoustics, Speech and Signal Processing (ICASSP), Taipei, Taiwan, April 2007, pp. 289-292.

Claims (120)

1. A unified time-domain/frequency-domain coding device for encoding input audio signals, comprising: A classifier that classifies the input sound signal into one of a plurality of sound signal categories, wherein the sound signal category includes an unclear signal type category, the unclear signal type category indicating that the nature of the input sound signal is unclear; A selector for encoding one of a plurality of encoding sub-patterns when the input audio signal is classified into the unclear signal type category; and A hybrid time-domain/frequency-domain encoder is used to encode the input audio signal using a selected coding sub-mode. 2. The time-domain/frequency-domain unified coding device according to claim 1, wherein the sound signal category includes speech, music, and an ambiguous signal type representing that the input sound signal is neither classified as speech nor music. 3. The unified time-domain/frequency-domain coding device according to claim 2, comprising: a frequency-domain encoder, used to encode the input sound signal when the classifier classifies the input sound signal as a music category. 4. The unified time-domain/frequency-domain coding apparatus according to claim 2 or 3, comprising: a time-domain encoder for encoding the input sound signal when the classifier classifies the input sound signal into a speech category. 5. The unified time-domain/frequency-domain coding apparatus according to any one of claims 1 to 4, wherein the selector selects the coding sub-mode in response to the bit rate for encoding the input audio signal and the characteristics of the input audio signal classified into the unclear signal type category. 6. The unified time-domain/frequency-domain coding apparatus according to any one of claims 1 to 5, wherein the coding sub-mode is identified by a corresponding sub-mode flag. 7. The unified time-domain/frequency-domain coding apparatus according to claim 5 or 6, wherein if (a) the bit rate available for encoding the input audio signal is not higher than a first given value and (b) the input audio signal is neither classified as speech nor as music, then the selector selects a backward coding submode that uses a conventional unified time-domain and frequency-domain coding model to encode the input audio signal. 8. The unified time-domain/frequency-domain coding apparatus according to any one of claims 5 to 7, wherein the selector selects a first coding sub-mode if a characteristic similar to "speech" is detected in the input sound signal. 9. The unified time-domain/frequency-domain coding apparatus according to claim 8, wherein the selector selects the first coding sub-mode if (a) the input sound signal is not classified as speech or music by the classifier and the bit rate available for encoding the input sound signal is higher than a second given value, (b) the probability that the input sound signal is music is not greater than a third given value, and (c) no time attack is detected in the current frame of the input sound signal. 10. The unified time-domain/frequency-domain coding apparatus according to any one of claims 5 to 9, wherein if a time attack is detected in the input audio signal, the selector selects a second coding sub-mode. 11. The unified time-domain/frequency-domain coding apparatus of claim 10, wherein the selector selects the second coding sub-mode if (a) the input sound signal is not classified as speech or music by the classifier and the bit rate available for encoding the input sound signal is higher than a fourth given value, (b) the probability that the input sound signal is music is not greater than a fifth given value, and (c) a time attack is detected in the current frame of the input sound signal. 12. The unified time-domain/frequency-domain coding apparatus according to any one of claims 5 to 11, wherein the selector selects a third coding sub-mode if a characteristic similar to "music" is detected in the input sound signal. 13. The unified time-domain/frequency-domain coding apparatus of claim 12, wherein the selector selects the third coding sub-mode if (a) the input sound signal is not classified as speech or music by the classifier and the bit rate available for encoding the input sound signal is higher than a sixth given value, and (b) the probability that the input sound signal is music is greater than a seventh given value. 14. The unified time-domain/frequency-domain coding apparatus according to any one of claims 1 to 13, wherein: - If a characteristic similar to "speech" is detected in the input sound signal, the selector selects the first encoding sub-mode; -If a time attack is detected in the input audio signal, the selector selects the second encoding sub-mode; If a characteristic similar to "music" is detected in the input sound signal, the selector selects the third encoding sub-mode. 15. The unified time-domain/frequency-domain coding apparatus of claim 14, wherein the selector (a) selects a given number of subframes frame-by-frame in the third coding sub-mode to encode the input audio signal, and (b) in the first coding sub-mode and the second coding sub-mode, selects a number of subframes less than the given number and based on the number of bit rates available for encoding the input audio signal. 16. A unified time-domain/frequency-domain coding device for encoding input audio signals, comprising: A classifier that classifies the input sound signal into one of a plurality of sound signal categories, wherein the sound signal category includes an unclear signal type category, the unclear signal type category indicating that the nature of the input sound signal is unclear; and A hybrid time-domain/frequency-domain encoder for encoding the input audio signal in response to the input audio signal being classified into the unclear signal type category; The hybrid time-domain/frequency-domain encoder includes a frequency band selector and a bit allocator for selecting the frequency band to be quantized and for allocating a bit budget available for quantization among the selected frequency bands. 17. The unified time-domain/frequency-domain coding apparatus of claim 16, wherein the hybrid time-domain/frequency-domain encoder includes a calculator of a difference vector between the frequency representations of the frequency-domain excitation contribution and the time-domain excitation contribution generated during hybrid time-domain/frequency-domain coding of the input audio signal, and wherein the frequency band is the frequency band of the difference vector. 18. The unified time-domain/frequency-domain coding apparatus of claim 16 or 17, wherein (a) the hybrid time-domain/frequency-domain encoder includes a cutoff frequency calculator, wherein above the cutoff frequency, the time-domain excitation contribution is not used for hybrid time-domain/frequency-domain coding of the input audio signal, and (b) the unified time-domain/frequency-domain coding apparatus includes a selector for encoding one of a plurality of coding sub-modes when the input audio signal is classified into the unclear signal type category, wherein the band selector and bit allocator select the band to be quantized and allocate the bit budget available for quantization in response to the cutoff frequency and the selected coding sub-mode. 19. The unified time-domain/frequency-domain coding apparatus according to claim 16, wherein: The hybrid time-domain/frequency-domain encoder includes: (a) a calculator for a cutoff frequency, wherein above the cutoff frequency, the time-domain excitation contribution is not used for hybrid time-domain/frequency-domain coding of the input audio signal; and (b) a calculator for a difference vector between the frequency-domain excitation contribution generated during the hybrid time-domain/frequency-domain coding of the input audio signal and the frequency representation of the time-domain excitation contribution; and The band selector and bit allocator include an estimator of a portion of the available bit budget that is frequency-quantized at lower frequencies of the difference vector as a function of the cutoff frequency. 20. The unified time-domain/frequency-domain coding apparatus of claim 19, comprising a selector for encoding one of a plurality of coding sub-modes when the input audio signal is classified into the unclear signal type category, wherein the estimator of a portion of the available bit budget adjusts the portion of the available bit budget based on the selected coding sub-mode. 21. The unified time-domain/frequency-domain coding apparatus of claim 20, wherein, when the selected coding sub-mode indicates the presence of a time attack in the input audio signal, the estimator of a portion of the available bit budget increases the portion of the available bit budget by a first given percentage, and when the selected coding sub-mode indicates the presence of musical characteristics in the input audio signal, decreases the portion of the available bit budget by a second given percentage. 22. The unified time-domain/frequency-domain coding apparatus of claim 17, comprising a selector for encoding one of a plurality of coding sub-modes when the input audio signal is classified into the unclear signal type category, wherein the band selector and the bit allocator comprise an estimator of the maximum number of bands of the difference vector to be quantized in relation to the selected coding sub-mode. 23. The unified time-domain/frequency-domain coding apparatus of claim 22, wherein the estimator for the maximum number of frequency bands (a) sets the maximum number of frequency bands of the difference vector to be quantized to a first value when selecting a first coding sub-mode in response to detecting a “speech”-like feature in the input audio signal, (b) sets the maximum number of frequency bands of the difference vector to be quantized to a second value when selecting a second coding sub-mode in response to detecting a time attack in the input audio signal, and (c) sets the maximum number of frequency bands of the difference vector to be quantized to a third value when selecting a third coding sub-mode in response to detecting a “musical”-like feature in the input audio signal. 24. The unified time-domain/frequency-domain coding apparatus of claim 22 or 23, wherein the estimator of the maximum number of frequency bands readjusts the maximum number of frequency bands of the difference vector to be quantized in response to a bit budget available for frequency quantization of the difference vector. 25. The unified time-domain/frequency-domain coding apparatus according to any one of claims 22 to 24, wherein the estimator of the maximum number of frequency bands further reduces the maximum number of frequency bands to be quantized in relation to the number of bits allocated to the intermediate and higher frequency bands of the difference vector. 26. The unified time-domain/frequency-domain coding apparatus according to any one of claims 19 to 21, wherein the band selector and bit allocator include a bit calculator, the bit calculator being allocated for frequency quantization of the lower frequency band of the difference vector in response to (a) a number of bits available for frequency quantization of the lower frequency of the difference vector and (b) a portion of a bit budget available for frequency quantization of the lower frequency. 27. The unified time-domain/frequency-domain coding apparatus of claim 26, wherein the bit calculator allocating for frequency quantization of the lower frequency band of the difference vector further responds to: (a) allocating a minimum number of bits for frequency quantization of the band, (b) allocating a number of bits for quantization of a first band following the lower frequency band, and (c) the difference between the estimated maximum number of bands to be quantized and the corrected, further reduced maximum number of bands to be quantized. 28. The unified time-domain/frequency-domain coding apparatus according to any one of claims 16 to 27, wherein the band selector and bit allocator include a band characterizer, the band characterizer being configured to (a) find bands with lower energy compared to their neighboring bands and mark the lower energy bands such that only a predetermined minimum number of bits can be allocated for frequency quantization of these lower energy bands, and (b) perform position sorting of other medium-energy bands and higher-energy bands. 29. The unified time-domain/frequency-domain coding device according to claim 28, wherein the frequency band characterizer sorts the mid-energy frequency band and the higher-energy frequency band in descending order of energy. 30. The unified time-domain/frequency-domain coding apparatus of claim 28 or 29, wherein the band selector and bit allocator include a final allocator of bits in the band to be quantized taking into account the frequency and energy of the band. 31. The unified time-domain/frequency-domain coding apparatus of claim 30, wherein, in order to perform frequency quantization on a lower frequency band, the final bit allocator linearly allocates bits allocated for frequency quantization of the lower frequency band, wherein a first percentage of bits is allocated to a first lowest frequency band, and a second percentage of bits is allocated to a last lower frequency band. 32. The unified time-domain/frequency-domain coding apparatus of claim 30 or 31, wherein the final allocator distributes the remaining bits allocated for frequency quantization of the difference vector according to a linear function to other mid-frequency bands and higher-frequency bands, but taking into account the frequency band energy characterization from the frequency band characterizer, such that more bits are allocated to higher-energy frequency bands and fewer bits are allocated to lower-energy frequency bands. 33. The unified time-domain/frequency-domain coding apparatus of claim 32, wherein the final allocator allocates any unallocated bits to the lower frequency band. 34. A unified time-domain/frequency-domain coding method for encoding input audio signals, comprising: The input sound signal is classified into one of a plurality of sound signal categories, wherein the sound signal category includes an unclear signal type category, the unclear signal type category indicating that the nature of the input sound signal is unclear; Select one of a plurality of encoding sub-modes for encoding the input audio signal when the input audio signal is classified into the unclear signal type category; and The input audio signal is mixed time-domain/frequency-domain encoded using the selected coding sub-mode. 35. The time-domain/frequency-domain unified coding method according to claim 34, wherein the sound signal category includes speech, music, and an ambiguous signal type representing that the input sound signal is neither classified as speech nor music. 36. The unified time-domain/frequency-domain coding method according to claim 35, comprising: performing frequency-domain coding on the input sound signal when the classifier classifies the input sound signal as a music category. 37. The unified time-domain/frequency-domain coding method according to claim 35 or 36, comprising: performing time-domain coding on the input sound signal when the classifier classifies the input sound signal into a speech category. 38. The unified time-domain/frequency-domain coding method according to any one of claims 34 to 37, wherein selecting one of a plurality of coding sub-modes includes: selecting the coding sub-mode in response to the bit rate used to encode the input audio signal and the characteristics of the input audio signal classified into the unclear signal type category. 39. The unified time-domain/frequency-domain coding method according to any one of claims 34 to 38, comprising identifying the coding sub-mode by a corresponding sub-mode flag. 40. The unified time-domain/frequency-domain coding method according to claim 38 or 39, wherein selecting one of a plurality of coding sub-modes includes: if (a) the bit rate available for encoding the input sound signal is not higher than a first given value and (b) the input sound signal is neither classified as speech nor as music, then a backward coding sub-mode that uses a conventional unified time-domain and frequency-domain coding model to encode the input sound signal is selected. 41. The unified time-domain/frequency-domain coding method according to any one of claims 38 to 40, wherein selecting one of a plurality of coding sub-modes includes: selecting a first coding sub-mode if a characteristic similar to "speech" is detected in the input sound signal. 42. The unified time-domain/frequency-domain coding method according to claim 41, wherein the first coding sub-mode is selected if (a) the input sound signal is not classified as speech or music and the bit rate available for encoding the input sound signal is higher than a second given value, (b) the probability that the input sound signal is music is not greater than a third given value, and (c) no time attack is detected in the current frame of the input sound signal. 43. The unified time-domain/frequency-domain coding method according to any one of claims 38 to 42, wherein selecting one of a plurality of coding sub-modes includes: selecting a second coding sub-mode if a time attack is detected in the input audio signal. 44. The unified time-domain/frequency-domain coding method according to claim 43, wherein the second coding sub-mode is selected if (a) the input sound signal is not classified as speech or music and the bit rate available for encoding the input sound signal is higher than a fourth given value, (b) the probability that the input sound signal is music is not greater than a fifth given value, and (c) a time attack is detected in the current frame of the input sound signal. 45. The unified time-domain/frequency-domain coding method according to any one of claims 38 to 44, wherein selecting one of a plurality of coding sub-modes includes: selecting a third coding sub-mode if a characteristic similar to "music" is detected in the input sound signal. 46. The unified time-domain/frequency-domain coding method according to claim 45, wherein the third coding sub-mode is selected if (a) the input sound signal is not classified as speech or music and the bit rate available for encoding the input sound signal is higher than a sixth given value, and (b) the probability that the input sound signal is music is greater than a seventh given value. 47. The unified time-domain/frequency-domain coding method according to any one of claims 34 to 46, wherein: - If a characteristic similar to "speech" is detected in the input sound signal, then the first encoding sub-mode is selected; -If a time attack is detected in the input audio signal, then the second encoding sub-mode is selected; - If a characteristic similar to "music" is detected in the input sound signal, then the third encoding sub-mode is selected. 48. The unified time-domain/frequency-domain coding method according to claim 47, wherein selecting one of a plurality of coding sub-modes includes: (a) in the third coding sub-mode, selecting a given number of subframes frame-by-frame to encode the input audio signal, and (b) in the first coding sub-mode and the second coding sub-mode, selecting a number of subframes less than the given number and based on the number of bit rates available for encoding the input audio signal. 49. A unified time-domain/frequency-domain coding method for encoding input audio signals, comprising: The input audio signal is classified into one of a plurality of audio signal categories, wherein the audio signal category includes an unclear signal type category, the unclear signal type category indicating that the nature of the input audio signal is unclear; and In response to the input audio signal being classified into the unclear signal type category, the input audio signal is subjected to mixed time-domain/frequency-domain coding; The mixed time-domain/frequency-domain coding of the input audio signal includes frequency band selection and bit allocation, used to select the frequency band to be quantized and to allocate a bit budget available for quantization among the selected frequency bands. 50. The unified time-domain/frequency-domain coding method according to claim 49, wherein performing mixed time-domain/frequency-domain coding on the input audio signal comprises: calculating a difference vector between the frequency representation of the frequency-domain excitation contribution and the frequency representation of the time-domain excitation contribution generated during the mixed time-domain/frequency-domain coding of the input audio signal, and wherein the frequency band is the frequency band of the difference vector. 51. The unified time-domain/frequency-domain coding method according to claim 49 or 50, wherein (a) performing mixed time-domain/frequency-domain coding on the input audio signal includes calculating a cutoff frequency, wherein above the cutoff frequency, the time-domain excitation contribution is not used for the mixed time-domain/frequency-domain coding of the input audio signal, and (b) the unified time-domain/frequency-domain coding method includes: selecting one of a plurality of coding sub-modes for encoding the input audio signal when the input audio signal is classified into the unclear signal type category, wherein the frequency band selection and bit allocation are in response to the cutoff frequency and the selected coding sub-mode to select the frequency band to be quantized and allocate the bit budget available for quantization. 52. The unified time-domain/frequency-domain coding method according to claim 49, wherein: Mixed time-domain/frequency-domain coding of the input audio signal includes: (a) calculating a cutoff frequency, wherein above the cutoff frequency, the time-domain excitation contribution is not used in the mixed time-domain/frequency-domain coding of the input audio signal; and (b) calculating the difference vector between the frequency-domain excitation contribution generated during the mixed time-domain/frequency-domain coding of the input audio signal and the frequency representation of the time-domain excitation contribution; and The frequency band selection and bit allocation include estimating a portion of the available bit budget for frequency quantization of the lower frequency of the difference vector as a function of the cutoff frequency. 53. The unified time-domain/frequency-domain coding method of claim 52, comprising: selecting one of a plurality of coding sub-modes for encoding the input audio signal when the input audio signal is classified into the unknown signal type category, wherein estimating a portion of the available bit budget includes adjusting the portion of the available bit budget based on the selected coding sub-mode. 54. The unified time-domain/frequency-domain coding method of claim 53, wherein estimating a portion of the available bit budget comprises: increasing the fraction of the available bit budget by a first given percentage if the selected coding sub-mode indicates the presence of a time attack in the input audio signal, and decreasing the fraction of the available bit budget by a second given percentage if the selected coding sub-mode indicates the presence of musical characteristics in the input audio signal. 55. The time-domain/frequency-domain unified coding method of claim 50, comprising: selecting one of a plurality of coding sub-modes for encoding the input audio signal when the input audio signal is classified into the unknown signal type category, wherein the frequency band selection and bit allocation includes estimating the maximum number of frequency bands of the difference vector to be quantized in relation to the selected coding sub-mode. 56. The unified time-domain/frequency-domain coding method according to claim 55, wherein estimating the maximum number of frequency bands of the difference vector comprises: (a) setting the maximum number of frequency bands of the difference vector to be quantized to a first value when a first coding sub-mode is selected in response to the detection of a “speech”-like feature in the input audio signal; (b) setting the maximum number of frequency bands of the difference vector to be quantized to a second value when a second coding sub-mode is selected in response to the detection of a time attack in the input audio signal; and (c) setting the maximum number of frequency bands of the difference vector to be quantized to a third value when a third coding sub-mode is selected in response to the detection of a “music”-like feature in the input audio signal. 57. The unified time-domain/frequency-domain coding method according to claim 55 or 56, wherein estimating the maximum number of frequency bands of the difference vector comprises: readjusting the maximum number of frequency bands of the difference vector to be quantized in response to a bit budget available for frequency quantization of the difference vector. 58. The unified time-domain/frequency-domain coding method according to any one of claims 55 to 57, wherein estimating the maximum number of frequency bands of the difference vector comprises: reducing the maximum number of frequency bands to be quantized in relation to the number of bits allocated to the mid-frequency band and higher-frequency band of the difference vector. 59. The unified time-domain/frequency-domain coding method according to any one of claims 52 to 54, wherein the frequency band selection and bit allocation comprises: calculating, in response to (a) the number of bits available for frequency quantization of the lower frequency of the difference vector, and (b) a portion of the bit budget available for frequency quantization of the lower frequency, the bits allocated for frequency quantization of the lower frequency band of the difference vector. 60. The unified time-domain/frequency-domain coding method of claim 59, wherein calculating the bits allocated for frequency quantization of the lower frequency band of the difference vector further responds to: (a) allocating a minimum number of bits for frequency quantization of the band, (b) allocating a number of bits for quantization of the first frequency band following the lower frequency band, and (c) the difference between the estimated maximum number of frequency bands to be quantized and the corrected, further reduced maximum number of frequency bands to be quantized. 61. The unified time-domain/frequency-domain coding method according to any one of claims 49 to 60, wherein the frequency band selection and bit allocation includes frequency band characterization for (a) finding frequency bands with lower energy compared to their neighboring frequency bands and marking the lower energy frequency bands such that only a predetermined minimum number of bits can be allocated for frequency quantization of these lower energy frequency bands, and (b) performing position sorting of other medium-energy frequency bands and higher energy frequency bands. 62. The unified time-domain/frequency-domain coding method according to claim 61, wherein the frequency band characterization includes sorting the medium-energy frequency band and the higher-energy frequency band in descending order of energy. 63. The unified time-domain/frequency-domain coding method according to claim 61 or 62, wherein the frequency band selection and bit allocation includes: a final allocation of bits in the frequency band quantized taking into account the frequency and energy of the frequency band. 64. The unified time-domain/frequency-domain coding method of claim 63, wherein, in order to perform frequency quantization on the lower frequency band, the final allocation of bits is linearly allocated to bits allocated for frequency quantization on the lower frequency, wherein a first percentage of bits is allocated to the first lowest frequency band, and a second percentage of bits is allocated to the last lower frequency band. 65. The unified time-domain/frequency-domain coding method according to claim 63 or 64, wherein the final allocation of the bits comprises: allocating the remaining bits to be used for frequency quantization of the difference vector to other mid-frequency bands and higher-frequency bands according to a linear function, but taking into account the frequency band energy characterization, such that more bits are allocated to higher-energy frequency bands and fewer bits are allocated to lower-energy frequency bands. 66. The unified time-domain/frequency-domain coding method of claim 65, wherein the final allocation of bits includes allocating any unallocated bits to the lower frequency band. 67. A unified time-domain/frequency-domain coding device for encoding input audio signals, comprising: At least one processor; and A memory coupled to the processor and storing non-transitory instructions that, when executed, cause the processor to implement: A classifier that classifies the input sound signal into one of a plurality of sound signal categories, wherein the sound signal category includes an unclear signal type category, the unclear signal type category indicating that the nature of the input sound signal is unclear; A selector for encoding one of a plurality of encoding sub-patterns when the input audio signal is classified into the ambiguous signal type category; and A hybrid time-domain/frequency-domain encoder is used to encode the input audio signal using a selected coding sub-mode. 68. A unified time-domain/frequency-domain coding device for encoding input audio signals, comprising: At least one processor; and A memory coupled to the processor and storing non-transitory instructions that, when executed, cause the processor to: The input sound signal is classified into one of a plurality of sound signal categories, wherein the sound signal category includes an unclear signal type category, the unclear signal type category indicating that the nature of the input sound signal is unclear; Select one of a plurality of encoding sub-modes for encoding the input audio signal when the input audio signal is classified into the unclear signal type category; and The input audio signal is mixed time-domain/frequency-domain encoded using the selected coding sub-mode. 69. A unified time-domain/frequency-domain coding device for encoding input audio signals, comprising: At least one processor; and A memory coupled to the processor and storing non-transitory instructions that, when executed, cause the processor to implement: A classifier that classifies the input sound signal into one of a plurality of sound signal categories, wherein the sound signal category includes an unclear signal type category, the unclear signal type category indicating that the nature of the input sound signal is unclear; and A hybrid time-domain/frequency-domain encoder for encoding the input audio signal in response to the input audio signal being classified into the unclear signal type category; The hybrid time-domain/frequency-domain encoder includes a frequency band selector and a bit allocator for selecting the frequency band to be quantized and for allocating a bit budget available for quantization among the selected frequency bands. 70. A unified time-domain/frequency-domain coding device for encoding input audio signals, comprising: At least one processor; and A memory coupled to the processor and storing non-transitory instructions that, when executed, cause the processor to: The input sound signal is classified into one of a plurality of sound signal categories, wherein the sound signal category includes an unclear signal type category, the unclear signal type category indicating that the nature of the input sound signal is unclear; In response to the input audio signal being classified into the unclear signal type category, the input audio signal is subjected to mixed time-domain/frequency-domain coding; and When performing mixed time-domain/frequency-domain coding on the input audio signal, the frequency band to be quantized is selected and the bit budget available for quantization is allocated between the selected frequency bands. 71. A sound signal decoder, comprising: A receiver of a bitstream that transmits information for reconstructing a hybrid time-domain/frequency-domain excitation, the hybrid time-domain/frequency-domain excitation representing an audio signal classified in an unclear signal type category, the unclear signal type category indicating that the nature of the audio signal is unclear, wherein the information includes one of a plurality of coding sub-patterns for encoding the audio signal classified in the unclear signal type category; A reconstructor that responds to a hybrid time-domain/frequency-domain excitation that includes information on coded sub-patterns for encoding the input audio signal transmitted in the bitstream; A converter that transforms the hybrid time-domain/frequency-domain excitation to the time domain; and A synthesis filter is used to filter the mixed time-domain/frequency-domain excitation converted to the time domain to produce a synthesized version of the sound signal. 72. The audio signal decoder of claim 71, wherein the encoded sub-mode is identified by a sub-mode flag in the bitstream. 73. The audio signal decoder according to claim 71 or 72, wherein the encoding sub-mode comprises: (a) a first encoding sub-mode in the case where the audio signal contains characteristics similar to "speech", (b) a second encoding sub-mode in the case where the audio signal contains time attacks, and (c) a third encoding sub-mode in the case where the audio signal contains characteristics similar to "music". 74. The audio signal decoder according to any one of claims 71 to 73, wherein the reconstructor recovers the frequency representation of the time-domain excitation contribution from the information transmitted in the bitstream, reconstructs the frequency-quantized difference vector between the frequency-domain excitation contribution and the frequency representation of the time-domain excitation contribution, and adds the frequency-quantized difference signal to the frequency representation of the time-domain excitation contribution to generate the hybrid time-domain/frequency-domain excitation. 75. A sound signal decoder, comprising: A receiver of a bitstream that transmits information that can be used to reconstruct a hybrid time-domain/frequency-domain excitation representing an audio signal, the audio signal being (a) classified into an ambiguous signal type category, the ambiguous signal type category indicating that the nature of the audio signal is ambiguous, and (b) encoded using (i) a frequency band selected for quantization and (ii) a bit budget available for quantization allocated between the frequency bands; A reconstructor in response to the information transmitted in the bitstream, wherein the reconstructor selects the frequency band for quantization and allocates a bit budget that can be used for quantization between the frequency bands; A converter that transforms the hybrid time-domain/frequency-domain excitation to the time domain; and A synthesis filter is used to filter the mixed time-domain/frequency-domain excitation converted to the time domain to produce a synthesized version of the sound signal. 76. The audio signal decoder of claim 75, wherein (a) the information from the bitstream includes a cutoff frequency and one of a plurality of coding sub-modes for encoding the audio signal classified into the unclear signal type category, wherein above the cutoff frequency, the time-domain excitation contribution is not used for the mixed time-domain/frequency-domain coding of the audio signal, and (b) the reconstructor, in response to the cutoff frequency and the coding sub-mode used, selects the quantized frequency band and allocates the bit budget available for dequantization between the selected frequency bands. 77. The audio signal decoder according to claim 75, wherein: Information from the bitstream includes a cutoff frequency, above which the time-domain excitation contribution is not used for the mixed time-domain/frequency-domain coding of the audio signal; and The reconstructor includes an estimator for frequency quantizing a portion of the available bit budget of the lower frequency of the difference vector between the frequency representation of the frequency domain excitation contribution and the frequency representation of the time domain excitation contribution generated during the mixed time-domain/frequency-domain coding of the audio signal. 78. The audio signal decoder of claim 77, wherein the information from the bitstream includes one of a plurality of coding sub-modes for encoding the audio signal, and wherein an estimator of a portion of the available bit budget adjusts the portion of the available bit budget based on the coding sub-mode used. 79. The audio signal decoder of claim 78, wherein, if the encoding sub-mode indicates the presence of a time attack in the audio signal, the estimator of a portion of the available bit budget increases the portion of the available bit budget by a first given percentage, and if the encoding sub-mode indicates the presence of musical features in the input audio signal, the estimator decreases the portion of the available bit budget by a second given percentage. 80. The audio signal decoder according to any one of claims 75 to 79, wherein the information from the bitstream includes one of a plurality of coding sub-modes for encoding the audio signal classified into the unclear signal type category, and wherein the reconstructor includes an estimator of the maximum number of frequency bands of the difference vector quantized in relation to the coding sub-mode used, the difference vector being determined between the frequency domain excitation contribution and the frequency representation of the time domain excitation contribution generated during mixed time-domain/frequency-domain coding of the audio signal. 81. The audio signal decoder of claim 80, wherein the estimator for the maximum number of frequency bands (a) sets the maximum number of frequency bands of the quantized difference vector to a first value when using a first coding sub-mode in response to detecting a “speech” characteristic in the audio signal, (b) sets the maximum number of frequency bands of the quantized difference vector to a second value when using a second coding sub-mode in response to detecting a time attack in the audio signal, and (c) sets the maximum number of frequency bands of the quantized difference vector to a third value when selecting a third coding sub-mode in response to detecting a “music” characteristic in the audio signal. 82. The audio signal decoder of claim 80 or 81, wherein the estimator of the maximum number of frequency bands readjusts the maximum number of frequency bands of the quantized difference vector in response to a bit budget available for frequency quantization of the difference vector. 83. The audio signal decoder according to any one of claims 80 to 82, wherein the estimator of the maximum number of frequency bands further reduces the maximum number of quantized frequency bands in relation to the number of bits of frequency quantization allocated to the mid-frequency band and higher-frequency band of the difference vector. 84. The audio signal decoder according to any one of claims 77 to 79, wherein the reconstructor includes a bit calculator, the bit calculator being allocated for frequency quantization of the lower frequency band of the difference vector in response to (a) a number of bits available for frequency quantization of the lower frequency of the difference vector, and (b) a portion of a bit budget available for frequency quantization of the lower frequency. 85. The audio signal decoder of claim 84, wherein the bit calculator allocating for frequency quantization of the lower frequency band of the difference vector is further responsive to (a) a minimum number of bits allocated to the quantized frequency band, (b) a number of bits allocated for quantization of a first frequency band following the lower frequency band, and (c) the difference between the estimated maximum number of quantized frequency bands and the corrected maximum number of further reduced quantized frequency bands. 86. The audio signal decoder according to any one of claims 75 to 85, wherein the reconstructor includes a frequency band characterizer, the frequency band characterizer being configured to (a) find frequency bands with lower energy compared to their neighboring frequency bands and mark the lower energy frequency bands such that only a predetermined minimum number of bits can be allocated for frequency quantization of these lower energy frequency bands, and (b) perform position sorting of other medium-energy frequency bands and higher energy frequency bands. 87. The audio signal decoder according to claim 86, wherein the frequency band characterizer sorts the mid-energy frequency band and the higher-energy frequency band in descending order of energy. 88. The audio signal decoder according to claim 86 or 87, wherein the reconstructor includes a final allocator of bits in the frequency band taking into account the quantization of the frequency and energy of the frequency band. 89. The audio signal decoder of claim 88, wherein, in order to perform frequency decoding on the lower frequency band, the final bit allocator linearly allocates the bits allocated to the lower frequency, wherein a first percentage of the bits is allocated to the first lowest frequency band, and a second percentage of the bits is allocated to the last lower frequency band. 90. The audio signal decoder of claim 88 or 89, wherein the final allocator distributes the remaining bits allocated for frequency dequantization of the difference vector to other mid-frequency bands and higher frequency bands according to a linear function, but takes into account the frequency band energy characterization from the frequency band characterizer, such that more bits are allocated to higher energy bands and fewer bits are allocated to lower energy bands. 91. The audio signal decoder of claim 90, wherein the final allocator assigns any unallocated bits to the lower frequency band. 92. The audio signal decoder according to any one of claims 75 to 91, wherein the reconstructor recovers the frequency representation of the time-domain excitation contribution from the information transmitted in the bitstream, reconstructs the frequency-quantized difference vector between the frequency-domain excitation contribution and the frequency representation of the time-domain excitation contribution from the information transmitted in the bitstream, and adds the frequency-quantized difference signal to the frequency representation of the time-domain excitation contribution to generate the hybrid time-domain/frequency-domain excitation. 93. The audio signal decoder of claim 92, wherein the reconstructor uses the frequency band selection and the allocation of the bit budget between the frequency bands to reconstruct the frequency quantization difference vector. 94. A method for decoding audio signals, comprising: Receive a bit stream that transmits information that can be used to reconstruct a mixed time-domain/frequency-domain excitation, the mixed time-domain/frequency-domain excitation representing a sound signal classified as an unclear signal type category, the unclear signal type category indicating that the nature of the sound signal is unclear, wherein the information includes one of a plurality of coding sub-patterns for encoding the sound signal classified as the unclear signal type category; In response to information including coded sub-patterns for encoding the input audio signal transmitted in the bitstream, the hybrid time-domain/frequency-domain excitation is reconstructed. Transform the hybrid time-domain/frequency-domain excitation to the time domain; and The mixed time-domain/frequency-domain excitation, converted to the time domain, is filtered by a synthesis filter to produce a synthesized version of the sound signal. 95. The audio signal decoding method according to claim 94, wherein the encoded sub-mode is identified by a sub-mode flag in the bitstream. 96. The sound signal decoding method according to claim 94 or 95, wherein the encoding sub-mode includes: (a) a first encoding sub-mode in the case where the sound signal contains characteristics similar to "speech", (b) a second encoding sub-mode in the case where the sound signal contains time attacks, and (c) a third encoding sub-mode in the case where the sound signal contains characteristics similar to "music". 97. The audio signal decoding method according to any one of claims 94 to 96, wherein reconstructing the hybrid time-domain/frequency-domain excitation comprises: recovering the frequency representation of the time-domain excitation contribution from the information transmitted in the bitstream; reconstructing the frequency-quantized difference vector between the frequency representations of the frequency-domain excitation contribution and the time-domain excitation contribution from the information transmitted in the bitstream; and adding the frequency-quantized difference signal to the frequency representation of the time-domain excitation contribution to generate the hybrid time-domain/frequency-domain excitation. 98. A method for decoding an audio signal, comprising: Receive a bit stream that transmits information that can be used to reconstruct a hybrid time-domain/frequency-domain excitation representing an audio signal, the audio signal being (a) classified into an ambiguous signal type category, the ambiguous signal type category indicating that the nature of the audio signal is ambiguous, and (b) encoded using (i) a frequency band selected for quantization and (ii) a bit budget available for quantization allocated between the frequency bands; The hybrid time-domain/frequency-domain excitation is reconstructed in response to the information transmitted in the bit stream, wherein reconstructing the hybrid time-domain/frequency-domain excitation includes selecting the frequency band for quantization and allocating a bit budget that can be used for quantization between the frequency bands; Transform the hybrid time-domain/frequency-domain excitation to the time domain; and The mixed time-domain/frequency-domain excitation, converted to the time domain, is filtered by a synthesis filter to produce a synthesized version of the sound signal. 99. The audio signal decoding method of claim 98, wherein (a) the information from the bitstream includes a cutoff frequency and one of a plurality of coding sub-modes for encoding the audio signal classified into the unclear signal type category, wherein above the cutoff frequency, the time-domain excitation contribution is not used for the mixed time-domain/frequency-domain coding of the audio signal, and (b) reconstructing the mixed time-domain/frequency-domain excitation includes: selecting a quantized frequency band in response to the cutoff frequency and the coding sub-mode used, and allocating the bit budget available for dequantization among the selected frequency bands. 100. The audio signal decoding method according to claim 98, wherein: The information from the bitstream includes a cutoff frequency, above which the time-domain excitation contribution is not used for the mixed time-domain/frequency-domain coding of the audio signal; and Reconstructing the hybrid time-domain/frequency-domain excitation includes: a portion of the available bit budget for frequency quantization of the lower frequency of the difference vector between the frequency-domain excitation contribution and the frequency representation of the time-domain excitation contribution generated during the hybrid time-domain/frequency-domain encoding of the audio signal. 101. The audio signal decoding method of claim 100, wherein the information from the bitstream includes one of a plurality of encoding sub-modes for encoding the audio signal, and wherein estimating a portion of the available bit budget includes adjusting the portion of the available bit budget based on the encoding sub-mode used. 102. The audio signal decoding method of claim 101, wherein estimating a portion of the available bit budget comprises: increasing the portion of the available bit budget by a first given percentage if the encoding sub-mode indicates the presence of a time attack in the audio signal, and decreasing the portion of the available bit budget by a second given percentage if the encoding sub-mode indicates the presence of musical features in the input audio signal. 103. The audio signal decoding method according to any one of claims 98 to 102, wherein the information from the bitstream includes one of a plurality of coding sub-modes for encoding the audio signal classified into the unclear signal type category, and wherein reconstructing the hybrid time-domain/frequency-domain excitation includes: estimating the maximum number of frequency bands of the difference vector quantized in relation to the coding sub-mode used, the difference vector being determined between the frequency domain excitation contribution and the frequency representation of the time domain excitation contribution generated during the hybrid time-domain/frequency-domain encoding of the audio signal. 104. The audio signal decoding method of claim 103, wherein estimating the maximum number of frequency bands comprises: (a) setting the maximum number of frequency bands of the quantized difference vector to a first value when using a first coding sub-mode in response to detecting a “speech” characteristic in the audio signal; (b) setting the maximum number of frequency bands of the quantized difference vector to a second value when using a second coding sub-mode in response to detecting a time attack in the audio signal; and (c) setting the maximum number of frequency bands of the quantized difference vector to a third value when selecting a third coding sub-mode in response to detecting a “music” characteristic in the audio signal. 105. The audio signal decoding method according to claim 103 or 104, wherein estimating the maximum number of frequency bands of the difference vector comprises: readjusting the maximum number of quantized frequency bands of the difference vector in response to a bit budget available for frequency quantization of the difference vector. 106. The audio signal decoding method according to any one of claims 103 to 105, wherein estimating the maximum number of frequency bands of the difference vector comprises: reducing the maximum number of quantized frequency bands in relation to the number of bits of frequency quantization allocated to the mid-frequency band and higher-frequency band of the difference vector. 107. The audio signal decoding method according to any one of claims 100 to 102, wherein reconstructing the hybrid time-domain/frequency-domain excitation comprises: calculating, in response to (a) the number of bits available for frequency quantization of the lower frequency of the difference vector, and (b) a portion of the bit budget available for frequency quantization of the lower frequency, the number of bits allocated for frequency quantization of the lower frequency band of the difference vector. 108. The audio signal decoding method of claim 107, wherein calculating the number of bits allocated for frequency quantization of the lower frequency band of the difference vector further responds to: (a) a minimum number of bits allocated to the quantized frequency band, (b) a number of bits allocated for quantization of a first frequency band following the lower frequency band, and (c) the difference between the estimated maximum number of quantized frequency bands and the corrected maximum number of further reduced quantized frequency bands. 109. The audio signal decoding method according to any one of claims 98 to 108, wherein reconstructing the hybrid time-domain/frequency-domain excitation includes characterizing the frequency bands for (a) finding frequency bands with lower energy compared to their neighboring frequency bands and marking the lower energy frequency bands such that only a predetermined minimum number of bits can be allocated for frequency quantization of these lower energy frequency bands, and (b) performing position sorting of other medium-energy frequency bands and higher energy frequency bands. 110. The audio signal decoding method according to claim 109, wherein characterizing the frequency band includes sorting the mid-energy frequency band and the higher-energy frequency band in descending order of energy. 111. The audio signal decoding method according to claim 109 or 110, wherein reconstructing the hybrid time-domain/frequency-domain excitation includes: final allocation of bits in the quantized frequency band taking into account the frequency and energy of the frequency band. 112. The audio signal decoding method according to claim 111, wherein the final allocation of bits comprises: in order to perform frequency dequantization on lower frequency bands, a linear allocation is made to the lower frequency bits, wherein a first percentage of bits is allocated to a first lowest frequency band, and a second percentage of bits is allocated to a last lower frequency band. 113. The audio signal decoding method according to claim 111 or 112, wherein the final allocation of the bits comprises: allocating the remaining bits allocated for frequency dequantization of the difference vector to other mid-frequency bands and higher frequency bands according to a linear function, but taking into account the energy characteristics of the frequency bands, such that more bits are allocated to higher energy frequency bands and fewer bits are allocated to lower energy frequency bands. 114. The audio signal decoding method of claim 113, wherein the final allocation includes allocating any unallocated bits to the lower frequency band. 115. The audio signal decoding method according to any one of claims 98 to 114, wherein reconstructing the hybrid time-domain/frequency-domain excitation comprises: recovering the frequency representation of the time-domain excitation contribution from the information transmitted in the bit stream, reconstructing the frequency-quantized difference vector between the frequency representation of the frequency-domain excitation contribution and the frequency representation of the time-domain excitation contribution from the information transmitted in the bit stream, and adding the frequency-quantized difference signal to the frequency representation of the time-domain excitation contribution to generate the hybrid time-domain/frequency-domain excitation. 116. The audio signal decoding method of claim 115, wherein reconstructing the hybrid time-domain/frequency-domain excitation comprises: reconstructing the difference vector of the frequency quantization using the frequency band selection and the allocation of the bit budget between the frequency bands. 117. A sound signal decoder, comprising: At least one processor; and A memory coupled to the processor and storing non-transitory instructions that, when executed, cause the processor to implement: A receiver of a bitstream that transmits information for reconstructing a hybrid time-domain/frequency-domain excitation, the hybrid time-domain/frequency-domain excitation representing a sound signal classified as an unclear signal type category, the unclear signal type category indicating that the nature of the sound signal is unclear, wherein the information includes one of a plurality of coding sub-patterns for encoding the sound signal classified as the unclear signal type category; A reconstructor that responds to a hybrid time-domain/frequency-domain excitation of information including coded sub-modes for encoding the input audio signal transmitted in the bitstream; A converter that transforms the hybrid time-domain/frequency-domain excitation to the time domain; and A synthesis filter is used to filter the mixed time-domain/frequency-domain excitation converted to the time domain to produce a synthesized version of the sound signal. 118. A sound signal decoder, comprising: At least one processor; and A memory coupled to the processor and storing non-transitory instructions that, when executed, cause the processor to: Receive a bitstream that conveys information that can be used to reconstruct a mixed time-domain/frequency-domain excitation, the mixed time-domain/frequency-domain excitation representing an audio signal classified as an unclear signal type category, the unclear signal type category indicating that the nature of the audio signal is unclear, wherein the information includes one of a plurality of coding sub-patterns for encoding the audio signal classified as the unclear signal type category; In response to information including coded sub-patterns for encoding the input audio signal transmitted in the bitstream, the hybrid time-domain/frequency-domain excitation is reconstructed. Transform the hybrid time-domain/frequency-domain excitation to the time domain; and The mixed time-domain/frequency-domain excitation, converted to the time domain, is filtered by a synthesis filter to produce a synthesized version of the sound signal. 119. A sound signal decoder, comprising: At least one processor; and A memory coupled to the processor and storing non-transitory instructions that, when executed, cause the processor to implement: A receiver of a bitstream that transmits information that can be used to reconstruct a hybrid time-domain/frequency-domain excitation representing an audio signal, the audio signal being (a) classified into an ambiguous signal type category, the ambiguous signal type category indicating that the nature of the audio signal is ambiguous, and (b) encoded using (i) a frequency band selected for quantization and (ii) a bit budget available for quantization allocated between the frequency bands; A reconstructor in response to the information transmitted in the bitstream, wherein the reconstructor selects the frequency band for quantization and allocates a bit budget that can be used for quantization between the frequency bands; A converter that transforms the hybrid time-domain/frequency-domain excitation to the time domain; and A synthesis filter is used to filter the mixed time-domain/frequency-domain excitation converted to the time domain to produce a synthesized version of the sound signal. 120. A sound signal decoder, comprising: At least one processor; and A memory coupled to the processor and storing non-transitory instructions that, when executed, cause the processor to: Receive a bit stream that transmits information that can be used to reconstruct a hybrid time-domain/frequency-domain excitation representing an audio signal, the audio signal being (a) classified into an ambiguous signal type category, the ambiguous signal type category indicating that the nature of the audio signal is ambiguous, and (b) encoded using (i) a frequency band selected for quantization and (ii) a bit budget available for quantization allocated between the frequency bands; The hybrid time-domain/frequency-domain excitation is reconstructed in response to the information transmitted in the bit stream, wherein the reconstruction selects the frequency band for quantization and the allocation of the bit budget that can be used for quantization between the frequency bands; Transform the hybrid time-domain/frequency-domain excitation to the time domain; and The mixed time-domain/frequency-domain excitation, converted to the time domain, is filtered by a synthesis filter to produce a synthesized version of the sound signal.