|Title of Invention||
AN APPARATUS FOR SELECTING AN ENCODING RATE FROM A PREDETERMINED SET OF ENCODING RATES FOR ENCODING A FRAME OF SPEECH
|Abstract||ABSTRACT It is an objective of the present invention to provide an optimized method of selection of the encoding mode that provides rate efficient coding of the input speech. It is a second objective of the present invention to identify and provide a means for generating a set of parameters ideally suited for this operational mode selection. Third, it is an objective of the present invention to provide identification of two separate conditions that allot/ low rate coding with minimal sacrifice to quality. The two conditions are the coding of unvoiced speech and the coding of temporally masked speech. It is a fourth objective of the present invention to provide a method for dynamically adjusting the average output data rate of the speech coder with marital impact on speech quality.|
The present invention relates to an apparatus for selecting an encoding rate from a predetermined set of encoding rates for encoding a frame of speech. More particularly, the present invention relates to a novel and improved apparels for performing variable rate code excited linear predictive (CELP) coding.
Transmission of voice by digital techniques has become widespread, particularly in long distance and digital radio telephone applications. This, in turn has created interest in determining the least amount of information which can be sent over the channel which maintains the perceived quality of the reconstructed speech. If specie is transmitted by simply sampling and digitizing, a data rate on the order of 64 kilobits per second (kbps) is required to achieve a speech quality of conventional analog telephone. However, through the use of speech analysis, followed by the appropriate coding, transmission and resynthesizes at the receiver, a significant reduction in the data rate can be achieved.
Devices which employ techniques to compress voiced speech by extracting parameters that relate to a model of human speech generation are typically called vocoder. Such devices are composed of an encoder, which the incoming speech to extract the relevant parameters, and a decoder, which resynthesizes the speech using the parameters which it receives over the transmission channel. In order to be accurate, the model must be constantly changing. Thus the speech is divided into blocks of time, or analysis frames, during which the parameters are calculated. The parameters are then updated for each new frame.
Of the various classes of speech coders the Code Excited Linear Predictive Coding (CELP), Stochastic Coding or Vector Excited Speech Coding are of one class. An example of a coding algorithm of this particular class is described in the paper " A 4.8 kbps Code Excited Linear Predictive Coder" by Thomas E Tremain et al.. Proceedings of the Mobile Satellite Conference, 1988.
The function of the vocoder is to compress the digitized speech signal
into a low bit rate signal by removing all of the natural redundancies inclement in
speech. Speech typically has short term redundancies due
primarily to the filtering operation of the vocal tract, and long term redundancies due to the excitation of the vocal tract by the vocal cords. In a CELP coder, these operations are modeled by two filters, a short term formant filter and a long term pitch filter. Once these redundancies are removed, the resulting residual signal can be modeled as white Gaussian noise, which also must be encoded. The basis of this technique is to compute the parameters of a filter, called the UPC filter, which performs short-term prediction of the speech waveform using a model of the human vocal tract. In addition, long-term effects, related to the pitch of the speech, are modeled by computing the parameters of a pitch filter, which essentially models the human vocal chords. Finally, these filters must be exited, and this is done by determining which one of a number of random excitation waveforms in a codebook results in the closest approximation to the original when the waveform excites the two filters mentioned above. Thus the transmitted parameters relate to three items (1) the IPC filter, (2) the pitch filter and (3) the codebook excitation
Although the use of hooding techniques further the objective in attempting to reduce the amount of information sent over the charnel while maintaining quality reconstructed speech, other techniques need be employed to achieve further reduction. One technique previously used to reduce the amount of information sent is voice activity gating. In this technique no information is transmitted during pauses in speech. Although this technique achieves the desired result of data reduction, it suffers from several deficiencies.
In many cases, the quality of speech is reduced due to clipping of the initial parts of word. Another problem with gating the charmel off during inactivity is that the system users perceive the lack of the background noise which normally accompanies speech and rate the quality of the channel as lower than a normal telephone call. A further problem with activity gating is that occasional sudden noises in the background may trigger the transmitter when no speech occurs, resulting in annoying bursts of noise at the receiver.
In an attempt to improve the quality of the synthesized speech in voice activity gating systems, synthesized comfort noise is added during the decoding process. Although some improvement in quality is achieved from adding comfort noise, it does not substantially improve the overall quality since the comfort noise does not model the actual background noise at the encoder.
A preferred technique to accomplish data compression, so as to result in a reduction of information that needs to be sent, is to perform variable rate vocoding. Since speech inherently contains periods of silence, i.e. pauses, the amount of data required to represent these periods can be 5 reduced. Variable rate vocoding most effectively exploits this fact by reducing the data rate for these periods of silence. A reduction in the data rate, as opposed to a complete halt in data transmission, for periods of silence overcomes the problems associated with voice activity gating while facilitating a reduction in transmitted information.
10 Copending U.S. Patent Application Serial No. 08/004,484, filed
January 14, 1993, entitled "Variable Rate Vocoder" and assigned to the assignee of the present invention and is incorporated by reference herein details a vocoding algorithm of the previously mentioned class of speech coders. Code Exited Linear Predictive Coding (CELP), Stochastic Coding or
15 Vector Exdted Speech Coding. The CELP technique by itself does provide a significant reduction in the amount of data necessary to represent speech in a manner that upon resynthesis results in high quality speech. As mentioned previously the vocoder parameters are updated for each frame. The vocoder detailed in the copending patent application provides a
20 variable output data rate by changing the frequency and precision of the model parameters.
The vocoding algorithm of the above mentioned patent application differs most markedly from the prior CELP techniques by producing a variable output data rate based on speech activity. The structiu-e is defined
25 so that the parameters are updated less often, or with less precision, during pauses in speech. This technique allows for an even greater decease in the amount of information to be transmitted. The phenomenon which is exploited to reduce the data rate is the voice activity factor, which is the average percentage of time a given speaker is actually talking during a
30 conversation. For typical two-way telephone conversations, the average data rate is reduced by a factor of 2 or more. During pauses in speech, only background noise is being coded by the vocoder. At these times, some of the parameters relating to the human vocal tract model need not be transmitted.
35 As mentioned previously a prior approach to limiting the amount of
information transmitted during silence is called voice activity gating, a technique in which no information is transmitted during moments of silence. On the receiving side the period may be filled in with synthesized "comfort noise". In contrast, a variable rate vocoder is continuously
transmitting data which, in the exemplary embodiment of the copending application, is at rates which range between approximately 8 kbps and 1 kbps. A vocoder which provides a continuous trar\smission of data eliminates the need for synthesized "comfort noise", with the coding of the background noise providing a more natural quality to the synthesized speech. The invention of the aforementioned patent application therefore provides a significant improvement in synthesized speech quality over that of voice activity gating by allowing a smcH)th transition between speech and background.
The vocoding algorithm of the above mentioned patent application enables short pauses in speech to be detected, a decrease in the effective voice activity factor is realized. Rate decisions can be made on a frame by frame basis with no hangover, so the data rate may be lowered for pauses in speech as short as the frame duration, typically 20 msec. Therefore pauses such as those between syllables may be captured. This technique decreases the voice activity factor beyond what has traditionally been considered, as not only long duration pauses between phrases, but also shorter pauses can be encoded at lower rates.
Since rale decisions are made on a frame basis, there is no clipping of the initial part of the word, such as in a voice activity gating system. Clipping of this nature occurs in voice activity gating system due to a delay between detection of the speech and a restart in transmission of data. Use of a rate decision based upon each frame results in speech where all trai\sitions have a natxiral sound.
With the vocoder always transmitting, the speaker's ambient background noise will continually be heard on the receiving end thereby yielding a more natural sound during speech pauses. The present invention thus provides a smooth transition to background noise. What the listener hears in the background during speech will not suddenly change to a synthesized comfort noise during pauses as in a voice activity gating system.
Since background noise is continually vocoded for transmission, interesting events in the background can be sent with full clarity. In cerjtain cases the interesting background noise may even be coded at the highest rate. Maximum rate coding may occur, for example, when there is someone talking loudly in the background, or if an ambulance drives by a user standing on a street corner. Constant or slowly varying background noise will, however, be encoded at low rates.
The use of variable rate vocoding has the promise of increasing the capacity of a Code Division Multiple Access (CDMA) based digital cellular telephone system by more than a factor of two. CDMA and variable rate vocoding are uniquely matched, since, with CDMA, the interference 5 between channels drops automatically as the rate of data transmission over any channel decreases. In contrast, consider systems in which transmission slots are assigned, such as TDMA or FDMA. In order for such a system to take advantage of any drop in the rate of data transmission, external intervention is required to coordinate the reassignment of unused slots to
10 other users. The inherent delay in such a scheme implies that the charmel may be reassigned only during long speech pauses. Therefore, full advantage cannot be taken of the voice activity factor. However, with external coordination, variable rate vocoding is useful in systems other than CDMA because of the other mentioned reasons.
15 In a CDMA system speech quality can be slightly degraded at times
when extra system capacity is desired. Abstractly speaking, the vocoder can be thought of as multiple vocoders all operating at different rates with different resultant speech qualities. Therefore the speech qualities can be mixed in order to further reduce the average rate of data transmission,
20 Initial experiments show that by mixing full and half rate vocoded speech, e.g. the maximum allowable data rate is varied on a frame by frame basis between 8 kbps and 4 kbps, the resulting speech has a quality which is better than half rate variable, 4 kbps maximum, but not as good as full rate variable, 8 kbps maximum.
25 It is well known that in most telephone conversations, only one
person talks at a time. As an additional function for full-duplex telephone links a rate interlock may be provided. If one direction of the link is transmitting at the highest transmission rate, then the other direction of the link is forced to transmit at the lowest rate. An interlock between the two
30 directions of the link can guarantee no greater than 50% average utilization of each direction of the link. However, when the channel is gated off, such as the case for a rate interlock in activity gating, there is no way for a listener to interrupt the talker to take over the talker role in the conversation. The vocoding method of the above mentioned patent application readily
35 provides the capability of an adaptive rate interlock by control signals which set the vocoding rate.
In the above mentioned patent application the vocoder operated at either full rate when speech is present or eighth rate when speech is not present. The operation of the vocoding algorithm at half and quarter rates
is reserved for special conditions of impacted capacity or when other data is to be transmitted in parallel with sp)eech data.
Copending U.S. Patent Application Serial No. 08/118,473, filed September 8, 1993, entitled "Method and Apparatus for Determining the 5 Transmission Data Rate in a Multi-User Communication System" and assigned to the assignee of the present invention and is incorporated by reference herein details a method by which a communication system in accordance with system capacity measurements limits the average data rate of frames encoded by a variable rate vocoder. The system reduces the
10 average data rate by forcing predetermined frames in a string of full rate frames to be coded at a lower rate, i.e. half rate. The problem with reducing the encoding rate for active speech frames in this fashion is that the limiting does not correspond to any characteristics of the input speech and so is not optimized for speech compression quality.
15 Also, in copending U.S. Patent Application Serial No. 07/984,602,
filed December 2, 1992, entitled "Improved Method for Determining Speech Encoding Rate in a Variable Rate Vocoder", now U.S. Patent NO. 5.341,456, issued August 23, 1994, and assigned to the assignee of the present invention and is incorporated by reference herein, a method for
20 distinguishing unvoiced speech from voiced speech is disclosed. The method disclosed examines the energy of the speech and the spectral tilt of the speech and uses the spectral tilt to distinguish unvoiced speech from background noise-Variable rate vocoders that vary the encoding rate based entirely on
25 the voice activity of the input speech fail to realize the compression efficiency of a variable rate coder that varies the encoding rate based on the complexity or irxformation content that is dynamically varying during active speech. By matching the encoding rates to the complexity of the input waveform more efficient speech coders can be built. Furthermore,
30 systems that seek to dynamically adjust the output data rate of the variable rate vocoders should vary the data rates in accordance with characteristics of the input speech to attain an optimal voice quality for a desired average data rate.
35 SUMMARY OF THE INVENTION
The present invention is a novel and improved method and apparatus for encoding active speech frames at a reduced data rate by encoding speech frames at rates between a predetermined maximum rate
and a predetermined minimum rate. The present Invention designates a set of active speech operation modes. In the exemplary embodiment of the present invention, there are four active speech operation modes, full rate speech, half rate speech, quarter rate unvoiced speech and quarter rate 5 voiced speech.
It is an objective of the present invention to provide an optimized method for selecting an encoding mode that provides rate efficient coding of the input speech. It is a second objective of the present invention to identify a set of parameters ideally suited for this operatioiukl mode selection and to
10 provide a means for generating this set of parameters. Third, it is an objective of the present invention to provide identification of two separate conditions that allow low rate coding with minimal sacrifice to quality. The two conditioris are the presence of unvoiced speech and the presence of temporally masked speech. It is a fourth objective of the present invention
15 to provide a method for dynamically adjusting the average output data rate of the speech coder with minimal impact on speech quality.
The present invention, provides a set of rate decision criteria referred to as mode measures. A first mode measure is the target matching signal to noise ratio (TMSNR) from the previous encoding frame, which provides
20 information on how well the synthesized speech matches the input speech or, in other words, how well the encoding model is performing. A second mode measure is the normalized autocorrelation function (NACF), which measures periodicity in the speech frame. A third mode measure is the zero crossings (ZC) parameter which is a computationally inexpei^sive method
25 for measuring high frequency content in an input speech frame. A fourth measure is the prediction gain differential (PGD) determines if the LPC model is maintaining its prediction efficiency. The fifth measure is the energy differential (ED) which compares the energy in the current frame to an average frame energy.
30 The exemplary embodiment of the vocoding algorithm of the present
invention uses the five mode measures enumerated above to select an
encoding mode for an active speech frame. The rate determination logic of
• the present invention compares the NACF agaiiut a first threshold value
and the ZC against a second threshold value to determine if the speech
35 should be coded as unvoiced quarter rate speech.
If it is determined that the active speech frame contains voiced speech, then the vocoder examines th^ parameter ED to determine if the speech frame should be coded as qiiarter rate voiced speech. If it is determined that the speech is not toibe coded at quarter rate, then the
vocoder tests if the speech can be coded at half rate. The vocoder tests the vahies of TMSNR, PGD and NACF to determine if the speech frame can be coded at half rate. If it is determined that tlie active speech frame cannot be coded at quarter or half rates, then the frame is coded at full rate.
It is further an objective to provide a method for dynamically changing threshold values in order to accommodate rate requirements. By varying one or more of the mode selection thresholds it is possible to increase or decrease the average data transmission rate. So by dynamically adjusting the threshold values an output rate can be adjusted.
Accordingly the present invention provides an apparatus for selecting an encoding rate from a predetermined set of encoding rates for encoding a frame of speech, having a pluralitv' of speech samples, comprising mode measurement logic, responsive to said speech samples and to a signal derived from said speech samples, for generating a set of parameters indicative of characteristics of said frame of speech; and rate determination logic for receiving said set of parameters and for selecting an encoding rate from said predetermined set of encoding rates using predetermined rate selection rules.
The features, objects, and advantages of the present invention will become more apparent from the detailed description set forth below when taken in conjunction with the drawings in which hke reference characters identify correspondingly throughout and wherein :
Figure 1 is a block diagram of the encoding rate determination apparatus of the present invention; and
Figure 2 is a flowchart illustrating the encoding rate selection process of the rate determination logic.
In the exemplary embodiment, speech frames of 160 speech samples are encoded. In the exemplary embodiment of the present invention, there are four data rates full rate, half rate, quarter rate and eighth rate. Full rate corresponds to an output data rate of 14.4 kbps. Half rate corresponds to an output data rate of 7.2 kbps. Quarter rate corresponds to data rate of 3.6 kbps. Eighth rate corresponds to an output data rate of 1.8 kbps, and is reserved for transmission during periods of silence.
It should be noted that the present invention relates only to the coding of active speech frames, frames that are detected to have speech present in them. The method for detecting the presence of speech is detailed in the aforementioned US Patent Application Serial Nos. 08/004,484 and 07/984,602.
Referring to figure 1, mode measurement element 12 determines values of five parameters used by rate determination logic 14 to select an encoding rate for the active speech frame. In the exemplary embodiment.
mode measurement element 12 determines five parameters which it
provides to rate deternxination logic 14. Based on the parameters provided
by mode measurement element 12, rate determination logic 14 selects an
encoding rate of full rate, half rate or quarter rate.
5 Rate determination logic 14 selects one of four encoding modes in
accordance with the five generated parameters. The four modes of encoding include full rate mode, half rate mode, quarter rate unvoiced mode and quarter rate voiced mode. Quarter rate voiced mode and quarter rate unvoiced mode provide data at the same rate but by means of different
10 encoding strategies. Half rate mode is used to code stationary, periodic, well modeled speech- Both quarter rate voiced, quarter rate unvoiced, and half rate modes take advantage of portions of speech that do not require high precision in the coding of the frame.
Quarter rate unvoiced mode is used in the coding of unvoiced
15 speech. Quarter rate voiced mode is used in the coding of temporally masked speech frames. Most CELP speech coders take advantage of simultaneous masking in which speech energy at a given frequency masks out noise energy at the same frequency and time making the noise inaudible. Variable rate speech coders can take advantage of temporal
20 masking in which low energy active speech frames are masked by preceding high energy speech frames of similar frequency content. Because the human ear is integrating energy over time in various frequency bands, low energy frames are time averaged with the high energy frames thus lowering the coding requirements for the low energy frames. Taking advantage of
25 this temporal masking auditory phenomena allows the variable rate speech coder to reduce the encoding rate during this mode of speech- This psychoacoustic phenomenon is detailed in Psychoacoustics by E. Zwicker and H- Fasti, pp. 56 -101.
Mode measurement element 12 receives four input signal with
30 which it generates the five mode parameters. The first signal that mode measurement element 12 receives is S(n) which is the uncoded input speech samples. In the exemplary embodiment, the speech samples are provided in frames containing 160 samples of speech- The speech frames that are provided to mode measurement element 12 all contain active
35 speech. During periods of silence, the active speech rate determination system of the present invention is inactive.
The second signal that mode measurement element 12 receives is the synthesized speech signal, S(n), which is the decoded speech from the encoder's decoder of the variable rate CELP coder. The encoder's decoder
decodes a frame of encoded speech for the purpose of updating filter parameters and memories in analysis by synthesis based CELP coder. The design of such decoders are well known in the art and are detailed in the above mentioned U.S. Patent Application Serial No. 08/004,484.
5 The third signal that mode measurement element 12 receives is the
formant residual signal e(n). The formant residual signal is the speech signal S(n) filtered by the linear prediction coding (LPC) filter of the CELP coder. The design of LPC filters and the filtering of signals by such filters is well known in the art and detailed in the above mentioned U.S. Patent
10 Application Serial No. 08/004,484. The fourth input to mode measurement element 12 is A(z) which are the filter tap values of the perceptual weighting filter of the associated CELP coder. The generation of the tap values, and filtering operation of a perceptual weighting filter are well known in the art and are detailed in U.S. Patent Application Serial No.
Target matching signal to noise ratio (SNR) computation element 2 receives the synthesized speech signal, S(n), the speech samples S(n), and a set of perceptual weighting filter tap values A(2). Target matching SNR computation element 2 provides a parameter, denoted TMSNR, which
20 indicates how well the speech model is tracking the input speech. Target matching SNR computation element 2 generates TMSNR in accordance with equation 1 below:
TMSNR = 10-log
159 ^ -
25 where the subscript w denotes that signal has been filtered by a perceptual weighting filter.
Note that this measure is computed for the previous frame of speech, while the NACF, PGD, ED, ZC are computed on the current frame of speech.
30 TMSNR is computed on the previous frame of speech since it is a function of the selected encoding rate and thus for computatioiul complexity reasons it is computed on the previous frame from the frame being encoded.
The design and implementation of perceptual weighting filters is well known in the art and is detailed in that aforementioned U.S. Patent
35 Application Serial No. 08/004,484. It should be noted that the perceptual
weighting is preferred to weight the perceptually significant features of the speech frame. However, it is envisioned that the measurement could be made without perceptually weighting the signals.
Normalized autocorrelation computation element 4 receives the
5 formant residual signal, e(n). The function of normalized autocorrelation
computation element 4 is lo provide an indication the periodicity of
samples in the speech frame. Normalized autocorrelation element 4
generates a parameter, denoted NACF in accordance with equation 2 below.
10 NACF= max ^^=^ • (2)
T€[20,120] ^1? 2, ,
It should be noted that the generation of this parameter requires memory of the formant residual signal from the encoding of the previous frame. This allows testing not only of the periodicity of the current frame, but also tests
15 the periodicity of the current frame with the previous frame.
The reason that in the preferred embodiment the formant residual signal, e(n), is used instead of the speech samples, S(n), which could be used, in generating NACF is to eliminate the interaction of the formants of the speech signal. Passing the speech signal though the formant filter serves to
20 flatten the speech envelope and thus whiterung the resxilting signal. It should be noted that the values of delay T in the exemplary embodiment correspond to pitch frequencies between 66 Hz and 400 Hz for a sampling frequency of 8000 samples per second. The pitch frequency for a given delay value T is calculated by equation 3 below:
f pitch = — / where fs is the sampling frequency. (3)
It should be noted that the frequency range can be extended or reduced
simply by selecting a different set of delay values. It should also be noted
30 that the present invention is equally applicable to any sampling frequencies.
Zero crossings counter 6 receives the speech samples S(n) and counts
the number of times the speech samples change sign. This is a
computationally inexpensive method of detecting high frequency
components in the speech signal. This counter can be implemented in software by a loop of the form:
5 for n=0,158 (5)
if (S(n)-S(n+1) The loop of equations 4-6 multiplies consecutive speech samples and tests if
the product is less than zero indicating that the sign between the two 10 consecutive samples differs. This assumes that there is no DC component
to the speech signal. It well known in the art how to remove DC
components from signals.
Prediction gain differential element 8 receives the speech signal S(n)
and the formant residual signal e(n). Prediction gain differential element 8 15 generates a parameter denoted PGD, which determines if the LPC model is
maintaining its prediction efficiency. Prediction gain differential element 8
generates the prediction gam, Pg, in accordance with equation 7 below:
I e^(n) n=0
The prediction gain of the present frame is then compared against the prediction gain of the previous frame in generating the output parameter PGD by equation 8 below:
25 PGD =10 log
— , where i denotes the frame number. (8)
In a preferred embodiment, prediction gain differential element 8 does not generate the prediction gain values Pg. In the generation of the LPC coefficients a byproduct of the Durbin's recursion is the prediction gain Pg so 30 no repetition of the computation is necessary.
Frame energy differential element 10 receives the speech samples s(n) of the present frame and computes the energy of the speech signal in the present frame in accordance with equation 9 below:
Ei = I S2(n) (9)
The energy of the present frame is compared to an average energy of previous frames Have- ^^ the exemplary embodiment, the average energy, 5 Have, is generated by a leaky integrator of the form:
Eave = ci'Eave + (l-a)'Ei, where 0 The factor, a, determines the range of frames that are relevant in the 10 computation. In the exemplary embodiment, the a is set to 0.8825 which provides a time constant of 8 frames. Frame energy differential element 10 then generates the parameter ED in accordance with equation 11 below:
The five parameters, TMSNR, NACF, ZC, PGD, and ED are provided to rate determination logic 14. Rate determination logic 14 selects an encoding rate for the next frame of samples in accordance with the parameters and a predetermined set of selection rules. Referring now to
20 Figure 2, a flow diagram illustrating the rate selection process of rate determination logic element 14 is shown.
The rate determination process begins in block 18. In block 20, the output of normalized autocorrelation element 4, NACF, is compared against a predetermined threshold value, THRl and the output of zero
25 crossings counter is compared against a second predetermined threshold, THR2. If NACF is less than THRl and ZC is greater than THR2, then the flow proceeds to block 22, which encodes the speech as quarter rate unvoiced. NACF being less than a predetermined threshold would indicate a lack of periodicity in the speech and ZC being greater than a
30 predetermined threshold would indicate high frequency component in the speech. The combination of these two conditions indicates that the frame contains unvoiced speech. In the exemplary embodiment THRl is 0.35 and THR2 is 50 zero aossing. If NACF is not less than THRl or ZC is not greater than THR2, then the flow proceeds to block 24.
35 In block 24, the output of frame energy differential element 10, ED, is
compared against a third threshold value, THR3, If ED is less than THR3,
then the current speech frame will be encoded as quarter rate voiced speech in block 26. If the energy difference between the current frame is lower than the average by a more than a threshold amount, then a condition of temporally masked speech is indicated. In the exemplary embodiment, 5 THR3 is -14dB. If ED does not exceed THR3 then the flow proceeds to block 28.
In block 28, the output of target matching SNR computation element 2, TMSNR, is compared to a fourth threshold value, THR4; the output of prediction gain differential element 8, PGD, is compared against a
10 fifth threshold value, THR5; and the output of normalized autocorrelation computation element 4, NACF, is compared against a sixth threshold value THR6. If TMSNR exceeds THR4; PGD is less than THR5; and NACF exceeds THR6, then the flow proceeds to block 30 and the speech is coded at half rate. TMSNR exceeding its threshold will indicate that the model and the speech
15 being modeled were matching well in the previous frame. The parameter PGD less than its predetermined threshold is indicative that the LPC model is maintaining its prediction efficiency. The parameter NACF exceeding its predetermined threshold indicates that the frame contains periodic speech that is periodic with the previous frame of speech.
20 In the exemplary embodiment, THR4 is initially set to 10 dB, THR5 is
set to -5 dB, and THR6 is set to 0.4. In block 28, if TMSNR does not exceed THR4, or PGD does not exceed THR5, or NACF does not exceed THR6, then the flow proceeds to block 32 and the cxirrent speech frame will be encoded at full rate.
25 By dynamically adjusting the threshold values an arbitrary overall
data rate can be achieved. The overall active speech average data rate, R, can be defined for an analysis window W active speech frames as:
Rf •# Rf frames + Rh # Rh frames + R„ •# Rqframes
R = -i i i^ ^ 3 2 , (12)
where Rf is the data rate for frames encoded at full rate, Rh is the data rate for frames encoded at half rate, Rq is the data rate for frames encoded at quarter rate, and W = #Rf frames + #Rh frames +#Rq frames.
By multiplying each of the encoding rates by the number of frames encoded at that rate and then dividing by the total number of frames in the sample an average data rate for the sampb of arfivp speech may be computed. It is
important to have a frame sample size, W, large enough to prevent a long
duration of unvoiced speech, such as drawn out "s" sounds from distorting
the average rate statistic. In the exemplary embodiment, the frame sample
size, W, for the calculation of the average rate is 400 frames.
5 The average data rate may be decreased by increasing the number of
frames encoded at full rate to be encoded at half rate and conversely the average data rate may be increased by increasing the number of frames encoded at half rate to be encoded at full rate. In a preferred embodiment the threshold that is adjusted to effect this change is THR4. In the
10 exemplary embodiment a histogram of the values of TSNR are stored. In the exemplary embodiment, the stored TMSNR values are quantized into values an integral number of decibels from the current value of THR4. By maintaining a histogram of this sort it can easily be estimated how many frames would have changed in the previous analysis block from being
15 encoded at full rate to being encoded at half rate were the THR4 to be decreased by an integral number of decibels. Conversely, an estimate of how many frames encoded at half rate would be encoded at full rate were the threshold to be increased by an integral number of decibels.
The equation for determining the number of frames that should
20 change from 1/2 rate frames to full rate frames is determined by the equation:
[target rate-average rate]-W Rf-Rh
25 where A is the number of frames encoded at half rate that should be encoded at full rate in order to attain the target rate, and W = #Rf frames + #Rh frames +#Rq frames.
TMSNRNEW=TMSNROLD + (the number of dB from TMSNROLD
30 to achieve A frame differences
defined in equation 13 above)
Note that the initial value of TMSNR is a function of the target rate desired. In an exemplary embodiment of a target rate of 8.7 Kbps, in a system with 35 Rf=14.4 kbps, Rf=7.2 kbps, Rq=3.6 kbps, the initial value of TMSNR is 10 dB. It should be noted that quantizing the TMSNR values to integral numbers for the distance from the threshold THR4 can easily be made finer such as
nail or quarter decibels or can be made coarser such as one and a half or two decibels.
It is envisioned that the target rate may either be stored in a memory element of rate determination logic element 14, in which case the target rate 5 would be a static value in accordance with which the THR4 value would be dynamically determined. In addition, to this initial target rate, it is envisioned that the communication system may transmit a rate command signal to the encoding rate Selection apparatus based upon current capacity conditions of the system.
10 The rate command signal could either specify the target rate or could
simply request an increase or decrease in the average rate. If the system were to specify the target rate, that rate would be used in determining the value of THR4 in accordance with equations 12 and 13. If the system specified only that the user should transmit at a higher or lower
15 transmission rate, then rate determination logic element 14 may respond by changing the THR4 value by a predetermined increment or may compute an incremental change in accordance with a predetermined incremental increase or decrease in rate.
Blocks 22 and 26 indicate a difference in the method of encoding
20 speech based upon whether the speech samples represent voiced or unvoiced speech. The unvoiced speech is speech in the form of fricatives and consonant sounds such as "i", "s", "sh", "t" and "z". Quarter rate voiced speech is temporally masked speech where a low volume speech frame follow a relatively high volume speech frame of similar frequency
25 content. The human ear cannot hear the fine points of the speech in the a low volume frame that follows a high volume frames so bits can be saved by encoding this speech at quarter rate.
In the exemplary embodiment of encoding unvoiced quarter rate speech, a speech frame is divided into four subframes. All that is
30 transmitted for each of the four subframes is a gain value G and the LPC filter coefficients A(z). In the exemplary embodiment, five bits are transmitted to represent the gain in each of each subframe. At a decoder, for each subframe, a codebook index is randomly selected. The randomly selected codebook vector is.multiphed by the traiismitted gain value and
35 passed through the LPC filter, A(z), to generate the synthesized unvoiced speech.
In the encoding of voiced quarter rate speech, a speech frame is divided into two subframes and the CELP coder determines a codebook index and gain for each of thp fwo subframes. In the exemplary
embodiment, five bits are allocated to indicating a codebook index and another five bits are allocated to specifying a corresponding gain value. In the exemplary embodiment, the codebook used for quarter rate voiced encoding is a subset of the vectors of the codebook used for half and full rate 5 encoding. In the exemplary embodiment, seven bits are used to specify a codebook index in the full and half rate encoding modes.
In Figure 1, the blocks may be implemented as structural blocks to perform the designated functions or the blocks may represent functions performed in programming of a digital signal processor (DSP) or an
10 application specific integrated circuit ASIC. The description of the functionality of the present invention would enable one of ordinary skill to implement the present invention in a DSP or an ASIC without undue experimentation.
The previous description of the preferred embodiments is provided
15 to enable any person skilled in the art to make or use the present invention. The various modifications to these embodiments will be readily apparent to those skilled in the art, and the generic principles defined herein may be applied to other embodiments without the use of the inventive faculty. Thus, the present invention is not intended to be limited to the
20 embodiments shown herein but is to be accorded the widest scope consistent with the principles and novel features disclosed herein.
1. An apparatus for selecting an encoding rate from a predetermined set of encoding rates for encoding a trade of speech, having a purity of speech samples, comprising mode measurement logic (12), responsive to said speech samples and to a signal derived from said speech samples, for generating a set of parameters indicative of characteristics of said frame of speech; and rate determination logic (14) for receiving said set of parameters and for selecting an encoding rate from said predetennined set of encoding rates using predetermined rate selection rules.
2. The apparatus as claimed in claim 1, wherein said mode measurement logic has means (2) to generate an encoding quality ratio indicative of a match between a previous frame of speech and synthesized speech derived there from.
3. The apparatus as claimed in claim 2, wherein said mode measurement logic has means (4) to generate a normalized autocorrelation measurement indicative of periodicity in said speech samples.
4. The apparatus as claimed in claim 2, wherein said mode measurement logic has means (6) to generate a zero crossings count indicative of a presence of high frequency components in said speech frame.
5. The apparatus as claimed in claim 2, wherein said mode measurement logic has means (8) to generate a prediction gain differential measurement indicative of a frame to frame stability of formants.
u. in apparatus as claimed in claim 2, wherein said mode measurement logic has means (10) to generate a frame energy differential measurement indicative of changes in energy between energy of said speech frame and an average frame energy.
7. The apparatus as claimed in claim 1, wherein said predetermined set of
encoding rates comprises full rate, half rate and quarter rate.
8. An apparatus for selecting an encoding rate from a predetennined set of
encoding rates for encoding a frame of speech, substantially as herein
described with reference to the accompanying drawings.
|Indian Patent Application Number||848/MAS/1995|
|PG Journal Number||30/2009|
|Date of Filing||07-Jul-1995|
|Name of Patentee||QUALCOMM INCORPORATED|
|Applicant Address||6455 LUSK BOULEVARD, SAN DIEGO CALIFORNIA 92121|
|PCT International Classification Number||N/A|
|PCT International Application Number||N/A|
|PCT International Filing date|