CN103474079A - Voice encoding method - Google Patents
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- CN103474079A CN103474079A CN2012102762176A CN201210276217A CN103474079A CN 103474079 A CN103474079 A CN 103474079A CN 2012102762176 A CN2012102762176 A CN 2012102762176A CN 201210276217 A CN201210276217 A CN 201210276217A CN 103474079 A CN103474079 A CN 103474079A
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Abstract
Provided is a voice encoding method. The voice encoding method mainly comprises two modules, namely encoders and decoders. High-frequency signals of input voice, after being processed via a high-frequency filter and a high-frequency encoder, are transmitted to a demultiplexer, while low-frequency signals of the input voice, after being processed via a low-frequency filter and a narrowband encoder, are transmitted to the demultiplexer. The signals, after passing through the demultiplexer, are transmitted to a multiplexer on the receiver side via a channel. The high-frequency signals output from the multiplexer are decoded by the high-frequency decoder and then are output; and the low-frequency signals, after being decoded by the narrowband decoder, are output together in integration with high and low frequency signals. The voice encoding method has a sampling rate of 6 kHz and a transmission code rate of 6.4 to 25.85 kbit/s. According to the voice encoding method, improvements are made on aspects of the removal of background noises and PLC processing, and thus significant improvement is made on the performance of the algorithm.
Description
Technical field
The present invention relates to the Software Coding field, be specifically related to a kind of voice coding method.
Background technology
The most convenient, the most effective and the most the most frequently used approach that voice are that people convey a message, it not only can transmit semantic information, can also pass on mood, attitude and other personal information.Critical positions due to voice in human communication, its analysis and application also become one of important research direction of information and signals process field.At present, speech processes mainly comprises following field: voice coding, and voice strengthen, and speech recognition and Speaker Identification also have phonetic synthesis.Therefore, voice and model thereof are goed deep into, multi-angular analysis, obtain characteristics of speech sounds and the tool that is for further processing is of great significance.Along with developing rapidly of internet, a kind of phone new business based on Internet-VoIP business appears.VoIP (Voice over IP, IP-based voice communication) also claims IP phone, is the novel digital transmission technology be based upon on Internet.The use of VoIP, can greatly reduce user's communication cost, and vigorously swelling of internet also promotes the fast development of voip technology.For making the voip network phone carry out reliably voice communication, must resolve two problems: the one, reduce coding bit rate under the prerequisite that guarantees certain speech quality; The 2nd, guarantee certain speech quality under the IP network environment.The former is the coding techniques that voice coding adopts, and same voice signal adopts different coded systems, and the bit rate after its coding is different, and the speech quality under the IP network environment is also different.G.723.1 the voice compressed encoding and decoding technology standard that VoIP mainly adopts has the ITU-T definition, G.729, G.729A etc., these technical standards are all used certain speed to be encoded to voice signal, and, for the unsettled characteristics of IP network environment, how using more effectively speech coding technology reduce coding bit rate under the prerequisite that guarantees certain speech quality is the required problem overcome.Current most of encoding and decoding speech systems of using are based on existing telephone bandwidth narrowband speech, this speech bandwidth is limited in about 300Hz-340Hz usually, the limit bandwidth of this inherence that sampling rate is 8kHz. tradition public switch telephone network (PSTN), restricted the further raising of speech communication quality, cause the naturalness at voice, the aspects such as music processing and some special processings are also unsatisfactory.Although this is unlikely to seriously to reduce the subjective quality of voice, in other many application, as amusement, voice broadcast service, teleconference, the occasions such as the communication of many coal and high-definition television, in the urgent need to higher-quality voice.
The research of speech coding technology has had the history of decades, is the period of a high speed development recently, and some metric systems have been made many speech coding standards, but most all for narrow generation voice.
Summary of the invention
In order to solve the deficiency of current existence, the invention provides a kind of practical voice coding method.
A kind of voice coding method, described encoding and decoding speech method mainly comprises two modules of encoder, high-frequency signal in the input voice transfers to coupler after high frequency filter and high frequency encoder processing, low frequency signal in the input voice transfers to coupler after low frequency filter and arrowband coder processes, after coupler signal by transmission the multiplexer to the other side, high-frequency signal output after the decoding of high frequency demoder from this multiplexer output, low frequency signal, after the decoding of arrowband demoder, is exported with the low-and high-frequency signal integration.
Preferably, the sampling rate of described voice coding method is 6kHz, and transmission code rate is 6.4-25.85kbit/s.
Preferably, the High frequency filter of described arrowband scrambler is 90Hz.
Preferably, while calculating in described arrowband scrambler, the LPC parameter is used LSF mean and quantize interpolation, and concrete grammar is:
At first by LSF(10) 3 little vectors (3,3,4) expression for vector, then use traditional vector quantization (VQ) to be quantized respectively it, for each little vector, at first design a code book, the size of these three code books is respectively:
Then use the code book designed to go to quantize, whole quantizing process has following three steps:
A. use the code book that size is 64 to go to quantize first three (1～3) LSF parameter.
B. use the code book that size is 128 to go to quantize three (4～6) LSF parameters subsequently.
C. use the code book that size is 128 to go to quantize four last (7～10) LSF parameters.
Wave filter corresponding to the LSF parameter after quantizing is:
Preferably, in the processing of described arrowband scrambler, also comprise and use analysis filter to calculate residue signal.
Preferably, in the processing of described arrowband scrambler, also comprise the coding of each frame sub-block is processed, the coding of frame comprises for the three kinds of sample block of encoding: 23/22 frame be left in 2 sub-blocks that comprise initial state; Sub-block before initial state; Sub-block after it is state, whole cataloged procedure is as follows:
1) use the encoded residual, information of code book decoding;
2) according to the newly-built memory pool of decoded result;
3) use filtrator to filter memory pool, and right to use heavy filtration device filter;
4) find the vector mated most with target in code book;
5) by constantly deducting the effect of selected vector, upgrade resulting weight target, repeat 4 and 5 twice;
6) calculate the energy lost because of coded residual information.
Preferably, in the processing of described arrowband scrambler, comprise the coding packing transmission to gained, the bits of needs transmission is divided three classes according to susceptibility: the most responsive is the first kind, and this class bits is placed on foremost; Inferior sensitivity be Equations of The Second Kind, this class bits is placed on the back of the first kind; The most insensitive is the 3rd class, and this class bits is placed on the last of each frame bit stream, and last at every frame bit stream means whether present frame loss with an empty frame prompt; Decoder module mainly contains following operation after receiving a voice packet:
If present frame is lost, step 1 to 5 is substituted with the PLC algorithm.
1) extract decoding parametric from data stream;
2) decoding LPC and carry out interpolation, this step is mainly to obtain the LP filtering parameter from quantize gained LSF vector;
3) rebuild initial state, our initial state decoding recovery after encode in this step;
4) to decoding rear gained residual, information put into storage pool;
5) the decode residual, information of each frame, repeat 4,5 steps until all frame pieces are all processed, whole process and be encoded to inverse process.In this process, the phenomenon of using PLC module meeting processed frame to lose.
6) residual, information is passed through to a boostfiltering device, the boostfiltering device can significantly reduce noise.
Preferably, the contents processing of described boostfiltering device comprises:
1) sample block of every two 80 sizes is carried out to the tone estimation;
2) find pitch-period-synchronous sequence n according to the tone of estimation in sample block k;
3) calculate level and smooth residue signal by front institute calling sequence;
4) whether effectively detect this residue signal;
5) use the related constraint condition to modify to strengthening signal, avoided enhancing;
6) this smooth rear signal and the residue signal do not strengthened is synthetic, in this step, the reasonable compensation enhancing is filtered the delay brought and is realized that filtering is synthetic, final information exchange is crossed to a high frequency filter and remove possible low frequency decoded signal.
Compared with prior art, advantage of the present invention is:
This cover error correcting capability of the present invention is strong, applicable to voice under arrowband are smooth, propagate, low encoding and decoding speech technology (the Internet Low Bit Rate Codec) technology to hardware requirement, removal at background noise, and PLC(packet loss concealment) the processing aspect improves, and makes the performance of algorithm that significant raising arranged.Can realize the voice transfer of the correct smoothness in the limited situation of the network bandwidth.Through experiment test, the error rate of this encoding and decoding technique is low, High frequency filter 90Hz adopt speed be 6kHz. at 2G, 2.5G, can realize the voice call of stable high-quality amount in 3G and WiFi situation, do not take massive band width and just can guarantee efficient voice transfer.
the accompanying drawing explanationthe schematic flow sheet that Fig. 1 is a kind of voice coding method of the present invention.
Embodiment
Shown in Fig. 1, a kind of voice coding method of the present invention, (the Internet Low Bit Rate Codec) technology based on the excitation of arrowband encoding and decoding technique, the sampling rate of 6kHz, transmission code rate 6.4-25.85kbit/s.In the removal of background noise, and PLC(packet loss concealment) the processing aspect improves, and makes the performance of algorithm that significant raising arranged.The encoding and decoding speech technology of this Project-developing mainly comprises demoder and two modules of scrambler, and coder module mainly contains following operation after receiving a phonetic entry that meets processing format:
Compare narrowband speech, this speech coding technology has been expanded under the limited bandwidth condition and has been improved voice quality, to input voice and pass through a high frequency filter to remove direct current component and low frequency noise, the contribution of HFS mainly is to have improved speech intelligibility, intelligibility and fricative texture.
Through experiment, the High frequency filter of scrambler is 90Hz.To calculate gained LPC parameter and use LSF mean and quantize interpolation, due to Linear Spectral Frequencies(LSF) than LPC, be more suitable for quantizing and interpolation.
(1) stability is strong
Because LSF need to pass to decoder end, consider the storage and computational problem, at first by LSF(10) vector with 3 little vectors (3,3,4), mean.Then use traditional vector quantization (VQ) to be quantized respectively it.For each little vector, at first design a code book.The size of these three code books is respectively:
Then we use the code book designed to go to quantize, and whole quantizing process has following three steps:
D. use the code book that size is 64 to go to quantize first three (1～3) LSF parameter.
E. use the code book that size is 128 to go to quantize three (4～6) LSF parameters subsequently.
F. use the code book that size is 128 to go to quantize four last (7～10) LSF parameters.
Through this quantizing process resulting LSF parameter corresponding unsettled LPC wave filter likely.We must detect the stability of LSF for this reason.If LSF is stable, we start it is looked into and controls.Whole Interpolation Process needs the LSF parameter (qlsf1, qlsf2) after initial LSF parameter (lsf1, lsf2) and quantification.Through the standardization interpolation, we can obtain two groups of wave filters.Wave filter corresponding to initial LSF parameter is:
Wave filter corresponding to the LSF parameter after quantizing is:
(2) use analysis filter to calculate residue signal
In this process, residue signal is processed and produced to sample sound by the LPC wave filter of gained after looking into.Select a sampling as initial state, LPC residue signal territory is transformed into to the voice domain of Weight with a weight filtrator, then detect therein two continuous sample sub-blocks with ceiling capacity, get (t1 by (4-4) formula, t2)=(2, the three rank semi-invariants that 4) can calculate their correspondences are respectively 0.14365 and 374.61875, the three rank accumulated amounts that this shows noise approach and zero, and three rank accumulated amount numerical value of voice signal are very large, in order to make its distribution more even, at first the residue signal of initial state is passed through to an all-pass filter.Use DPCM to quantize acquired results, whole process is as follows:
(3) for each frame, all in code database, searched for
Each frame sub-block is encoded.Code book in this step will be used for the three kinds of sample block of encoding: remaining 23/22 frame in 2 sub-blocks that comprise initial state; Sub-block before initial state; Sub-block after it is state, whole cataloged procedure is as follows:
1) use the encoded residual, information of code book decoding.
2) according to the newly-built memory pool of decoded result.
3) use filtrator to filter memory pool, and right to use heavy filtration device filter.
4) find the vector mated most with target in code book.
5), by constantly deducting the effect of selected vector, upgrade resulting weight target.Repeat 4 and 5 twice.
6) calculate the energy lost because of coded residual information.
(4) by gained coding packing transmission
The bits of needs transmission is divided three classes according to susceptibility: the most responsive is the first kind (class 1), and this class bits is placed on foremost; Inferior sensitivity be Equations of The Second Kind (class 2), this class bits is placed on the back of the first kind; The most insensitive is the 3rd class (class 3), and this class bits is placed on the last of each frame bit stream, and last at every frame bit stream means whether present frame loss with an empty frame prompt.Decoder module mainly contains following operation after receiving a voice packet:
If present frame is lost, step 1 to 5 is substituted with the PLC algorithm.
1. extract decoding parametric from data stream.
2. LPC and carry out interpolation decodes.This step is mainly to obtain the LP filtering parameter from quantize gained LSF vector.
3. reconstruction initial state.In this step, our the initial state decoding after encoding recovers.
4. to decoding rear gained residual, information put into storage pool.
5. the decode residual, information of each frame.Repeat 4,5 steps until all frame pieces are all processed.Whole process and be encoded to inverse process.In this process, we use PLC(packet loss concealment) phenomenon can processed frame lost of module.
6. residual, information is passed through to a boostfiltering device, the boostfiltering device can significantly reduce noise.Compare and traditional amplifier the modification excitation signal that the boostfiltering device that we use only can be slight.This has been avoided crossing the enhancing phenomenon.
1) sample block of every two 80 sizes is carried out to the tone estimation.
2) find pitch-period-synchronous sequence n according to the tone of estimation in sample block k.
3) calculate level and smooth residue signal by front institute calling sequence
4) whether effectively detect this residue signal.
5) use the related constraint condition to modify to strengthening signal, avoided enhancing.
6) this smooth rear signal and the residue signal do not strengthened is synthetic.
Synthetic residual, information, in this step, our reasonable compensation strengthens and filters the delay brought and realize that filtering is synthetic, final information exchange is crossed to a high frequency filter and remove possible noise, and to obtain noise be the low frequency decoded signal to indication here.
Above-described embodiment just is to allow the one of ordinary skilled in the art can understand content of the present invention and implement according to this for technical conceive of the present invention and characteristics being described, its objective is, can not limit the scope of the invention with this.Variation or the modification of every equivalence that the essence of content has been done according to the present invention, all should be encompassed in protection scope of the present invention.
Claims (8)
1. a voice coding method, described encoding and decoding speech method mainly comprises two modules of encoder, high-frequency signal in the input voice transfers to coupler after high frequency filter and high frequency encoder processing, low frequency signal in the input voice transfers to coupler after low frequency filter and arrowband coder processes, after coupler signal by transmission the multiplexer to the other side, high-frequency signal output after the decoding of high frequency demoder from this multiplexer output, low frequency signal, after the decoding of arrowband demoder, is exported with the low-and high-frequency signal integration.
2. voice coding method according to claim 1, is characterized in that, the sampling rate of described voice coding method is 6kHz, and transmission code rate is 6.4-25.85kbit/s.
3. voice coding method according to claim 1, is characterized in that, the High frequency filter of described arrowband scrambler is 90Hz.
4. voice coding method according to claim 1, is characterized in that, while calculating in described arrowband scrambler, the LPC parameter is used LSF mean and quantize interpolation, and concrete grammar is:
At first by LSF(10) 3 little vectors (3,3,4) expression for vector, then use vector quantization (VQ) to be quantized respectively it, for each little vector, at first design a code book, the size of these three code books is respectively:
At first by LSF(10) 3 little vectors (3,3,4) expression for vector, then use vector quantization (VQ) to be quantized respectively it, for each little vector, at first design a code book, the size of these three code books is respectively:
Then use the code book designed to go to quantize, whole quantizing process has following three steps:
A. use the code book that size is 64 to go to quantize first three (1～3) LSF parameter;
B. use the code book that size is 128 to go to quantize three (4～6) LSF parameters subsequently;
C. use the code book that size is 128 to go to quantize four last (7～10) LSF parameters;
Detect the stability of LSF, if LSF is stable, just start it is carried out to interpolation; Whole Interpolation Process needs the LSF parameter (qlsf1, qlsf2) after initial LSF parameter (lsf1, lsf2) and quantification, through the standardization interpolation, can obtain two groups of wave filters.
5. voice coding method according to claim 1, is characterized in that, in the processing of described arrowband scrambler, also comprises and use analysis filter to calculate residue signal.
6. according to the described voice coding method of claim 4 or 5, it is characterized in that, in the processing of described arrowband scrambler, comprise the coding of each frame sub-block is processed, the coding of frame comprises for the three kinds of sample block of encoding: 23/22 frame be left in 2 sub-blocks that comprise initial state; Sub-block before initial state; Sub-block after it is state, whole cataloged procedure is as follows:
1) use the encoded residual, information of code book decoding;
2) according to the newly-built memory pool of decoded result;
3) use filtrator to filter memory pool, and right to use heavy filtration device filter;
4) find the vector mated most with target in code book;
5) by constantly deducting the effect of selected vector, upgrade resulting weight target, repeat 4 and 5 twice;
6) calculate the energy lost because of coded residual information.
7. voice coding method according to claim 6, it is characterized in that, in the processing of described arrowband scrambler, also comprise the coding packing transmission to gained, the bits of needs transmission is divided three classes according to susceptibility: the most responsive is the first kind, and this class bits is placed on foremost; Inferior sensitivity be Equations of The Second Kind, this class bits is placed on the back of the first kind; The most insensitive is the 3rd class, and this class bits is placed on the last of each frame bit stream, and last at every frame bit stream means whether present frame loss with an empty frame prompt; Decoder module mainly contains following operation after receiving a voice packet:
If present frame is lost, step 1 to 5 is substituted with the PLC algorithm;
1) extract decoding parametric from data stream;
2) decoding LPC and carry out interpolation, this step is mainly to obtain the LP filtering parameter from quantize gained LSF vector;
3) rebuild initial state, our initial state decoding recovery after encode in this step;
4) to decoding rear gained residual, information put into storage pool;
5) the decode residual, information of each frame, repeat 4,5 steps until all frame pieces are all processed, whole process and be encoded to inverse process, and in this process, the phenomenon of using the PLC module can processed frame to lose;
6) residual, information is passed through to a boostfiltering device, the boostfiltering device can significantly reduce noise.
8. voice coding method according to claim 7, is characterized in that, the contents processing of described boostfiltering device comprises:
1) sample block of every two 80 sizes is carried out to the tone estimation;
2) find pitch-period-synchronous sequence n according to the tone of estimation in sample block k;
3) calculate level and smooth residue signal by front institute calling sequence;
4) whether effectively detect this residue signal;
5) use the related constraint condition to modify to strengthening signal, avoided enhancing;
6) this smooth rear signal and the residue signal do not strengthened is synthetic, in this step, the reasonable compensation enhancing is filtered the delay brought and is realized that filtering is synthetic, final information exchange is crossed to a high frequency filter and remove possible low frequency decoded signal.
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| CN108510990A (en) * | 2018-07-04 | 2018-09-07 | 百度在线网络技术（北京）有限公司 | Audio recognition method, device, user equipment and storage medium |
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Application publication date: 20131225 |