CN112133318A - Digital voice coding device - Google Patents
Digital voice coding device Download PDFInfo
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- CN112133318A CN112133318A CN202011018258.6A CN202011018258A CN112133318A CN 112133318 A CN112133318 A CN 112133318A CN 202011018258 A CN202011018258 A CN 202011018258A CN 112133318 A CN112133318 A CN 112133318A
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- 230000015572 biosynthetic process Effects 0.000 claims abstract description 4
- 238000000695 excitation spectrum Methods 0.000 claims abstract description 4
- 238000009499 grossing Methods 0.000 claims abstract description 4
- 238000003786 synthesis reaction Methods 0.000 claims abstract description 4
- 230000006835 compression Effects 0.000 abstract description 4
- 238000007906 compression Methods 0.000 abstract description 4
- 230000006837 decompression Effects 0.000 abstract description 2
- 238000000034 method Methods 0.000 description 7
- 238000004891 communication Methods 0.000 description 6
- 230000003044 adaptive effect Effects 0.000 description 2
- 238000010586 diagram Methods 0.000 description 2
- 230000000694 effects Effects 0.000 description 2
- 238000010295 mobile communication Methods 0.000 description 2
- 101000622137 Homo sapiens P-selectin Proteins 0.000 description 1
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- 101000873420 Simian virus 40 SV40 early leader protein Proteins 0.000 description 1
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
- G10L19/0204—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/26—Pre-filtering or post-filtering
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- Physics & Mathematics (AREA)
- Computational Linguistics (AREA)
- Signal Processing (AREA)
- Health & Medical Sciences (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- Spectroscopy & Molecular Physics (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
Abstract
The invention discloses a digital voice coding device, which comprises a coder and a decoder, wherein the coder comprises a pulse code modulation module, a linear programming analysis module, a margin frame energy module, an excitation spectrum analysis module, a fundamental tone estimation and smoothing algorithm module and a molecular band voiced and unvoiced decision module; the decoder includes: the device comprises a sine sum generator, a noise generator, a band-pass filter bank, a synthesis filter module and a post-filter module. The invention solves the problems that the residual signal of the very low-rate speech coding is difficult to compress, the speech quality after decompression is too low, the compression algorithm is too complex and the engineering is difficult to realize.
Description
Technical Field
The invention relates to a voice coding technology, in particular to a digital voice coding device.
Background
Speech coding is currently of considerable importance in communication systems, and to a large extent determines the received speech quality and the system capacity. Conventionally, speech coding techniques can be divided into three categories, waveform coding, parametric coding and hybrid coding.
Waveform coding is to directly convert a time domain signal into a digital code in an attempt to maintain a reconstructed speech waveform in the waveform shape of an original speech signal. The basic principle of waveform coding is to sample analog speech at a certain rate on the time axis, then quantize amplitude samples hierarchically, and represent them with codes. Decoding is the reverse process, and the received digital sequence is decoded and filtered to recover an analog signal. Pulse Code Modulation (PCM) and Delta Modulation (DM), as well as various modified Adaptive Delta Modulation (ADM), adaptive differential coding (ADPCM), etc., thereof, are all waveform coding techniques. However, for network communication with increasingly strained bandwidth resources, the coding method is obviously not suitable, because the communication rate of waveform coding is generally above the rate of 16 Kbit/s. The parameter coding is used for extracting and coding the characteristic parameters of the signal envelope, and at a decoding end, the original voice signal is reconstructed through the characteristic parameters and the residual error, so that the compression rate is high. Typical speech compression coding methods are LPC-10, MELP, SELP, etc. Parameter coding is widely used in mobile communication at present because the communication rate can reach 1.2-2.4 Kbit/s. Hybrid coding, i.e. a process of coding using two or more coding methods simultaneously. Since the hybrid coding combines the high quality of waveform coding and the low data rate of parametric coding, it achieves better effect in practical application. Typical hybrid coding schemes are MPLPC, KPELPC, CELP, etc.
In mobile communication systems, broadband resources are at a premium. Low bit rate speech coding provides one approach to this problem. On the premise that the encoder can transmit high-quality voice, if the bit rate is lower, more channels of high-quality voice can be transmitted in a certain broadband. Speech coding is source coding that converts an analog speech signal into a digital signal for transmission in a channel. The purpose of speech coding is to transmit speech of the highest possible quality while occupying as little communication capacity as possible while maintaining a certain algorithm complexity and communication delay.
Disclosure of Invention
The present invention has been made to solve the above-mentioned problems occurring in the prior art by providing a digital speech encoding apparatus.
The technical problem solved by the invention is realized by adopting the following technical scheme:
a digital voice coding device comprises a coder and a decoder, wherein the coder comprises a pulse code modulation module, a linear programming analysis module, a margin frame energy module, an excitation spectrum analysis module, a fundamental tone estimation and smoothing algorithm module and a molecular band unvoiced and voiced sound judgment module; the decoder includes: the device comprises a sine sum generator, a noise generator, a band-pass filter bank, a synthesis filter module and a post-filter module.
The invention solves the problems that the residual signal of the very low-rate speech coding is difficult to compress, the speech quality after decompression is too low, the compression algorithm is too complex and the engineering is difficult to realize.
Drawings
FIG. 1 is a block diagram of an encoder structure of a digital speech encoder according to the present invention.
FIG. 2 is a block diagram of a decoder structure of a digital speech coder according to the present invention.
Detailed Description
In order to make the technical means, the creation characteristics, the achievement purposes and the effects of the invention easy to understand, the invention is further described with the specific embodiments.
As shown in fig. 1 and 2, a digital speech encoding apparatus includes: an encoder and a decoder, wherein the encoder receives an original voice, encodes the original voice, and transmits; the decoder receives the data sent by the encoder, decodes the data and synthesizes voice; the encoder comprises a pulse code modulation module 02, a linear programming analysis module 04, a margin frame energy module 05, an excitation spectrum analysis module 06, a fundamental tone estimation and smoothing algorithm module 07 and a molecular band unvoiced and voiced sound judgment module 08; the decoder includes: a sine sum generator 09, a noise generator 10, a band pass filter bank 11, a synthesis filter module 12, a post filter module 13.
The foregoing shows and describes the general principles and broad features of the present invention and advantages thereof. It will be understood by those skilled in the art that the present invention is not limited to the embodiments described above, which are described in the specification and illustrated only to illustrate the principle of the present invention, but that various changes and modifications may be made therein without departing from the spirit and scope of the present invention, which fall within the scope of the invention as claimed. The scope of the invention is defined by the appended claims and equivalents thereof.
Claims (1)
1. A digital speech encoding apparatus comprising an encoder and a decoder, characterized in that: the encoder comprises a pulse code modulation module, a linear programming analysis module, a margin frame energy module, an excitation spectrum analysis module, a fundamental tone estimation and smoothing algorithm module and a molecular band unvoiced and voiced sound judgment module; the decoder includes: the device comprises a sine sum generator, a noise generator, a band-pass filter bank, a synthesis filter module and a post-filter module.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN202011018258.6A CN112133318A (en) | 2020-09-24 | 2020-09-24 | Digital voice coding device |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN202011018258.6A CN112133318A (en) | 2020-09-24 | 2020-09-24 | Digital voice coding device |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CN112133318A true CN112133318A (en) | 2020-12-25 |
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Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN202011018258.6A Withdrawn CN112133318A (en) | 2020-09-24 | 2020-09-24 | Digital voice coding device |
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| Country | Link |
|---|---|
| CN (1) | CN112133318A (en) |
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2020
- 2020-09-24 CN CN202011018258.6A patent/CN112133318A/en not_active Withdrawn
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Legal Events
| Date | Code | Title | Description |
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| PB01 | Publication | ||
| PB01 | Publication | ||
| WW01 | Invention patent application withdrawn after publication |
Application publication date: 20201225 |
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| WW01 | Invention patent application withdrawn after publication |