CN101800608B - Adaptive differential pulse-code modulation-demodulation system and method - Google Patents
Adaptive differential pulse-code modulation-demodulation system and method Download PDFInfo
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- CN101800608B CN101800608B CN200910008673.0A CN200910008673A CN101800608B CN 101800608 B CN101800608 B CN 101800608B CN 200910008673 A CN200910008673 A CN 200910008673A CN 101800608 B CN101800608 B CN 101800608B
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Abstract
The invention discloses an adaptive differential pulse-code modulation-demodulation system and an adaptive differential pulse-code modulation-demodulation method, which can fulfil the main aims of simplifying the hardware complexity of a decoding end and network transmission quantity and improving sound quality. The system comprises an encoding and modulating module and a decoding demodulating module. The method comprises the following steps: receiving an analogue sound input signal, converting the signal into a data packet and transmitting the data packet to a communication network, wherein the data packet at least comprises a plurality of adaptive differential pulse-code modulating quantity bit data, and an initial value and a scale factor corresponding to the adaptive differential pulse-code modulating quantity bit data; and receiving the data packet, and processing the adaptive differential pulse-code modulating quantity bit data, the initial value and the scale factor in the data packet so as to restore an analogue sound output signal.
Description
Technical field
The present invention relates to a kind of adaptive differential pulse-code modulation-demodulation system and method, espespecially a kind of adaptive differential pulse-code modulation-demodulation system and method being applied on telecommunication network.
Background technology
While using pulse-code modulation (hereinafter to be referred as PCM) to produce digitized voice data, conventionally need to mention two parameters.First parameter is sampling rate (Sample Rate), is the quantity that the speech waveform in each second is cut apart.Sampling rate is higher, and the quality of voice is certain also better.Conventionally sampling rate is that 8000 hertz (Hz) have good effect, but if produce the sound quality as CD, sampling rate need to arrive 44100Hz.Second parameter is sample size (Sample size), is to express speech data with how many Wei Wei units.Because the data that store in PCM are amplitude sizes of speech waveform, if we take a byte (8bits), be unit, maximum and minimum difference of vibration only have 255.If but we change that to take two bytes (16bits) be unit, maximum and minimum amplitude can differ to 65535, voice quality is at this time certainly also better.But the data that represent voice with more position, system own not only needs more internal memory, also needs to have good digital signal processor (DSP) to arrange in pairs or groups and just can accomplish simultaneously.And when these speech digit data will transmit through network, sizable frequency range that also certainly will account for, causes the serious burden of Internet resources.
Therefore, designer is just by differential pulse-code modulation (differential pulse-code modulation, hereinafter to be referred as DPCM) concept introduction, due to DPCM record is current value and the difference value of previous value, therefore carry out comparison with simple pulse-code modulation, DPCM data volume that method produces on average approximately can be down to 25% of original data volume.And adaptive differential pulse code modulation becomes (adaptive differentialpulse-code modulation, hereinafter to be referred as ADPCM) be the distortion of DPCM, what data volume can be recompressed is less, can allow like this frequency range of transmission channel increase, and this technology is described in detail in ITU-TG.726 standard, do not repeat them here.But, in ITU-T G.726 in standard, at the receiving terminal hardware that is applied to above-mentioned adaptive differential pulse code modulation change method, need more complicated design successfully to decompress and restore voice data compress, so will cause expensive hardware cost and cause product competitiveness downslide.And how to develop a new technological means, effectively solve the above-mentioned disappearance of commonly using means, be development main purpose of the present invention.
Summary of the invention
Technical problem to be solved by this invention is to provide a kind of adaptive differential pulse-code modulation-demodulation system and method, can reach simplified decoding end hardware complexity, transmission volume and improve the main purpose of sound quality.
In order to solve above technical problem, the invention provides following technical scheme:
The invention provides a kind of adaptive differential pulse-code modulation-demodulation system, comprise: a coding modulating module, be coupled to a telecommunication network, it is to receive a simulated sound input signal to convert a data packet to and send to this telecommunication network, and this Datagram at least includes a plurality of adaptive differential pulse code modulation variable bit data and corresponding to an initial value and a scale factor of these adaptive differential pulse code modulation variable bit data; An and decoding solution modulating module, be coupled to this telecommunication network, it is to receive this Datagram, and processes according to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this Datagram, and then restores a simulated sound output signal.
According to above-mentioned conception, adaptive differential pulse-code modulation-demodulation system of the present invention, wherein this coding modulating module comprises: a quantifying unit, its be with one fixedly sampling frequency this simulated sound input signal is sampled and is encoded into the digital sampling value of a plurality of uncompressed; One Difference Calculation unit, is electrically connected on this quantifying unit, and it is the digital sampling value of these uncompressed is subtracted each other with an initial value respectively and correspondingly obtain a plurality of digital difference; One scale factor maker, is electrically connected on this Difference Calculation unit, and it is according to the difference between the maximum sampling value in the digital sampling value of this initial value and uncompressed in the scheduled time, to estimate a scale factor of corresponding this data packet; One compression unit, is electrically connected on this Difference Calculation unit, in order to these digital difference are compressed into these adaptive differential pulse code modulation variable bit data; One package generator, is electrically connected on this compression unit and this scale factor maker, in order to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this scheduled time are packaged into this data packet; And a conveyer, be electrically connected on this package generator, in order to this data packet is sent to this telecommunication network.
According to above-mentioned conception, adaptive differential pulse-code modulation-demodulation system of the present invention, wherein this decoding solution modulating module comprises: a receiver, in order to receive this data packet; One package plug-in reader, is electrically connected on this receiver, in order to understand these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this data packet; One decompression unit, is electrically connected on this package plug-in reader, these adaptive differential pulse code modulation variable bit data that read out in order to take out this package plug-in reader solution; One scale factor adjuster, is electrically connected on this package plug-in reader, this initial value and this scale factor that in order to take out this package plug-in reader solution, read out; One Difference Calculation unit, is electrically connected on this decompression unit and this scale factor adjuster, in order to carry out computing according to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor, and then obtains the digital sampling value of a plurality of decompressions; And a de-quantization unit, be electrically connected on this Difference Calculation unit, in order to receive the digital sampling value of these decompressions, carry out de-quantization processing and restore this simulated sound output signal.
According to above-mentioned conception, adaptive differential pulse-code modulation-demodulation system of the present invention, wherein this telecommunication network is internet, LAN or mobile phone network.
Another aspect of the present invention provides a kind of adaptive differential pulse code modulation to become-separate modulating method, comprise the following step: receive a simulated sound input signal and convert a data packet to, and in this data packet, including a plurality of adaptive differential pulse code modulation variable bit data, an initial value and a scale factor; This Datagram is sent to a telecommunication network; And receive this Datagram through this telecommunication network, and process according to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this Datagram, and then restore a simulated sound output signal.
According to above-mentioned conception, adaptive differential pulse code modulation of the present invention becomes-separates modulating method, and wherein this step that receives this simulated sound input signal comprises: with one fixedly sampling frequency this simulated sound input signal is sampled and is encoded into the digital sampling value of a plurality of uncompressed; The digital sampling value of these uncompressed is subtracted each other with an initial value respectively and correspondingly obtain a plurality of digital difference; According to the difference between the maximum sampling value in the digital sampling value of this initial value and uncompressed in the scheduled time, estimate a scale factor of corresponding this data packet; These numerical digit difference modulations are become to these adaptive differential pulse code modulation variable bit data; And these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this scheduled time are packaged into this data packet.
According to above-mentioned conception, adaptive differential pulse code modulation of the present invention becomes-separates modulating method, and wherein this step that restores this simulated sound output signal comprises: understand these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this Datagram; Take out these adaptive differential pulse code modulation variable bit data that this package plug-in reader solution reads out; Take out this initial value and this scale factor that this package plug-in reader solution reads out; According to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor, carry out computing, and then obtain the digital sampling value of a plurality of uncompressed; And the digital sampling value that receives these uncompressed carries out de-quantization processing and restores this simulated sound output signal.
Adaptive differential pulse-code modulation-demodulation system and method that the present invention adopts, package is directly delivered to decoding end by initial value and scale factor, effectively improve the disappearance of prior art, effectively reduce hardware complexity and the cost that adaptive differential pulse code modulation becomes communication system, and can be widely used in the fields such as the real-time transfer voice (Real-time audio transmission overTCP/IP network) of carrying out on internet, LAN or mobile phone network or the networking telephone (Voice over Internet Protocol is called for short VoIP).
Accompanying drawing explanation
Fig. 1 (a) becomes for adaptive differential pulse code modulation the sampling computing schematic diagram that method (ADPCM) develops out for the present invention.
For the present invention develops out, a kind of adaptive differential pulse code modulation becomes-separates the flow chart of steps of modulating method to Fig. 1 (b).
Fig. 2 is a kind of adaptive differential pulse-code modulation-demodulation system function block schematic diagram that the present invention develops out.
Fig. 3 is the preferred embodiment schematic diagram of data packet form of the present invention.
[primary clustering symbol description]
Graphic middle each the comprised assembly of the present invention lists as follows:
Decoding solution modulating module 92 receivers 26
Package plug-in reader 27 decompression units 28
Scale factor adjuster 29 Difference Calculation unit 30
De-quantization unit 31
Embodiment
Fig. 1 (a) becomes for adaptive differential pulse code modulation the sampling computing schematic diagram that method (ADPCM) develops out for the present invention, wherein p (n) representative is with the digital sampling value of a uncompressed that fixedly sampling frequency obtains after one sound signal waveform S is sampled, and x (n) representative is through calculus of differences and data compression process digital sampling value later, and x (0)=p0, x (1)=(p1-p0)/sf, the rest may be inferred for x (2)=(p2-p1)/sf etc., wherein p0 is called basic pulse code modulation value (base PCM value), and sf is called scale factor (scalefactor).
And be to reduce the required hardware cost of data receiving terminal, a kind of adaptive differential pulse code modulation that the present invention proposes as shown in Fig. 1 (b) becomes-separates modulating method, the method comprises the following step: step S11: receive a simulated sound input signal and convert a data packet to, and in this data packet, include a plurality of adaptive differential pulse code modulation variable bit data, and independent field is deposited an initial value and a scale factor corresponding to these adaptive differential pulse code modulation variable bit data; Step S12: Datagram is sent to telecommunication network; And step S13: see through telecommunication network and receive data packet, and process according to adaptive differential pulse code modulation variable bit data, initial value and scale factor in Datagram, and then restore a simulated sound output signal.
Fig. 2 is the adaptive differential pulse-code modulation-demodulation system function block schematic diagram of preferred embodiment of the present invention, comprise two major parts, first part is a coding modulating module 91, signal is connected on this telecommunication network 2, it is to receive a simulated sound input signal S (raw input) to convert plurality of data package to and send to telecommunication network 2, and wherein at least includes a plurality of adaptive differential pulse code modulation variable bit data in a Datagram and corresponding to an initial value and a scale factor of these adaptive differential pulse code modulation variable bit data.As for the second part, it is a decoding solution modulating module 92, signal is connected to telecommunication network 2, it is to receive to see through the data packet that telecommunication network 2 sends, and process according to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this Datagram, and then restore a simulated sound output signal S '.
And coding modulating module 91 comprises scale factor maker 20, quantifying unit 21, Difference Calculation unit 22, compression unit 23, package generator 24 and conveyer 25, wherein simulated sound input signal S sends to after quantifying unit 21, quantifying unit 21 is that this simulated sound input signal is carried out to quantification treatment (quantization), with one fixedly sampling frequency analog signal is sampled and is encoded into the digital sampling value (p (n) as shown in Fig. 1 (a)) of a plurality of uncompressed, and Difference Calculation unit 22 is the digital sampling value of these uncompressed is subtracted each other with an initial value (p (0) as shown in Fig. 1 (a)) respectively and correspondingly obtain a plurality of digital difference.The scale factor maker 20 that is electrically connected on Difference Calculation unit 22 just can estimate according to the digital difference between the maximum sampling value in the digital sampling value of this initial value and uncompressed in the scheduled time scale factor of corresponding this data packet.Then, then 23 of compression units that are electrically connected on this Difference Calculation unit 22 deliver to package generator 24 in order to a plurality of digital difference are compressed into these adaptive differential pulse code modulation variable bit data, as for being electrically connected on the package generator 24 of this compression unit 23 with this scale factor maker 20,, in order to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor of (or predetermined quantity) in this scheduled time are packaged into this data packet, then see through conveyer 25 and send to telecommunication network 2.
By above-mentioned explanation, can be found out, for reducing the long-range hardware cost of telecommunication network 2, the present embodiment is provided with scale factor maker 20, preferably, according to the digital difference between the maximum sampling value in the digital sampling value of this initial value and uncompressed in this scheduled time, estimate a scale factor of corresponding this data packet, for example, adaptive differential pulse code modulation variable bit data can only reach with 4 bit tables, but in this package, variable quantity between initial value and maximum need to be used 6 positions while expressing, scale factor is just made as 4, so just numerical data length all in this package all can be reduced into 4 positions.And owing to needing scale factor in the long-range decompression process of telecommunication network 2, therefore facility is found out the scale factor in each data packet with scale factor maker 20, be put in data packet and send together with initial value.
Refer to Fig. 3, it is for the preferred embodiment schematic diagram of data packet form of the present invention, it is with RTP (Real-time Transport Protocol, hereinafter to be referred as RTP) describe for example, therefore data packet form system follows the regulation of RTP, and RTP is a kind of host-host protocol that point-to-point transmission service is provided, be used for being supported in Unicast and Multicast network service and carry out transfer of data, sound program request (audio-on-demand), VOD (video on demand), Internet telephony (Internet telephone) and video conference (video conferencing) all can be applied.The package specification of RTP is not formulated standard to the compressed format of sound and TV, so it can be used to transmit sound and the video files of common format, can certainly be used for transmitting sound and the video files of proprietary format storage.The sound being generated by multimedia application and video files are encapsulated in RTP package, and each RTP package can be encapsulated in UDP (User Datagram Protocol, user's data flow communications protocol) in message section, and then be encapsulated in IP package.In gauge outfit (Header) due to RTP package, be provided with the gauge outfit that a certain bits represents this package and whether be attached with in addition an extension field, in this embodiment, this certain bits in the gauge outfit of RTP package (Header) is inserted to 1, the gauge outfit that represents this RTP package has extension field, the extension field of 32 of take is example, front 16 in order to put the initial value of this package, then 16 in order to put scale factor.Therefore,, in the package that package generator 24 of the present invention produces, except the numerical data of fixed qty, also comprise initial value and the scale factor relevant to these numerical datas.
Return to see again Fig. 2, after including the numerical data of fixed qty and the RTP package of the initial value relevant to these numerical datas and scale factor and being received by the receiver 26 in long-range decoding solution modulating module 92, just send package plug-in reader 27 solutions to and read these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this Datagram.In this embodiment, this certain bits in the gauge outfit of RTP package (Header) is 1, therefore package plug-in reader 27 is untied and is passed to decompression unit 28 except the adaptive differential pulse code modulation variable bit data (x (n) as shown in Fig. 1 (a)) that package main body is transmitted, this certain bits in response gauge outfit is 1, also can read the extension field contents shown in Fig. 3, and then must return this initial value and scale factor.And scale factor adjuster 29 just can take out this initial value and this scale factor that this package plug-in reader solution reads out, as for 28 these adaptive differential pulse code modulation variable bit data that read out in order to take out these package plug-in reader 27 solutions of decompression unit, then again by be electrically connected on this decompression unit 28 with the Difference Calculation unit 30 of this scale factor adjuster 29 according to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor are carried out decompression operation, and then obtain the digital sampling value (p (n) as shown in Fig. 1 (a)) of a plurality of decompressions, last by the digital sampling value of 31 pairs of these the received decompressions in de-quantization unit, carry out de-quantization processing (de-quantization) again, finally restore the simulated sound output signal S ' (raw output) that approaches simulated sound input signal.
And from said process, because package of the present invention is directly delivered to decoding end by initial value and scale factor, therefore the present invention can effectively improve the disappearance of prior art, reaches simplified decoding end hardware complexity, transmission volume and improves the main purpose of sound quality.
In sum, the present invention effectively reduces hardware complexity and the cost that adaptive differential pulse code modulation becomes communication system, and can be widely used in the fields such as the real-time transfer voice (Real-t ime audio transmission over TCP/IP network) of carrying out on internet, LAN or mobile phone network or the networking telephone (Voice over Internet Protocol is called for short VoIP).As for RTP (Real-time Transport Protocol; hereinafter to be referred as RTP) also can use other network package communications protocol instead; therefore the present invention must be thought and is to modify as all by the personage Ren Shi craftsman who has the knack of this skill, neither disengaging protection scope of the present invention.
Claims (12)
1. an adaptive differential pulse-code modulation-demodulation system, is characterized in that, comprises:
One coding modulating module, be coupled to a telecommunication network, it is to receive a simulated sound input signal to convert a data packet to and send to this telecommunication network, and this data packet at least includes a plurality of adaptive differential pulse code modulation variable bit data and corresponding to an initial value and a scale factor of these adaptive differential pulse code modulation variable bit data, this data packet is a RTP data packet, wherein the certain bits in package gauge outfit is 1, represent that this data packet has an extension field, and this initial value and this scale factor system are arranged in this extension field, and
One decoding solution modulating module, be coupled to this telecommunication network, it is to receive this data packet, and processes according to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this data packet, and then restores a simulated sound output signal.
2. adaptive differential pulse-code modulation-demodulation system as claimed in claim 1, is characterized in that, this coding modulating module comprises:
One quantifying unit, its be with one fixedly sampling frequency this simulated sound input signal is sampled and is encoded into the digital sampling value of a plurality of uncompressed;
One Difference Calculation unit, is electrically connected on this quantifying unit, and it is the digital sampling value of these uncompressed is subtracted each other with an initial value respectively and correspondingly obtain a plurality of digital difference;
One scale factor maker, is electrically connected on this Difference Calculation unit, and it is according to the difference between the maximum sampling value in the digital sampling value of this initial value and uncompressed in the scheduled time, to estimate a scale factor of corresponding this data packet;
One compression unit, is electrically connected on this Difference Calculation unit, in order to these digital difference are compressed into these adaptive differential pulse code modulation variable bit data;
Package generator, is electrically connected on this compression unit and this scale factor maker, in order to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this scheduled time are packaged into this data packet; And
One conveyer, is electrically connected on this package generator, in order to this data packet is sent to this telecommunication network.
3. adaptive differential pulse-code modulation-demodulation system as claimed in claim 1, is characterized in that, this decoding solution modulating module comprises:
One receiver, in order to receive this data packet;
One package plug-in reader, is electrically connected on this receiver, in order to understand these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this data packet;
One decompression unit, is electrically connected on this package plug-in reader, these adaptive differential pulse code modulation variable bit data that read out in order to take out this package plug-in reader solution;
One scale factor adjuster, is electrically connected on this package plug-in reader, this initial value and this scale factor that in order to take out this package plug-in reader solution, read out;
One Difference Calculation unit, is electrically connected on this decompression unit and this scale factor adjuster, in order to carry out computing according to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor, and then obtains the digital sampling value of a plurality of decompressions; And
One de-quantization unit, is electrically connected on this Difference Calculation unit, in order to receive the digital sampling value of these decompressions, carries out de-quantization processing and restores this simulated sound output signal.
4. adaptive differential pulse-code modulation-demodulation system as claimed in claim 1, is characterized in that, this extension field is 32.
5. adaptive differential pulse-code modulation-demodulation system as claimed in claim 4, is characterized in that, this initial value and this scale factor occupy respectively 16 in this extension field.
6. adaptive differential pulse-code modulation-demodulation system as claimed in claim 1, is characterized in that, this telecommunication network is internet, LAN or mobile phone network.
7. adaptive differential pulse code modulation becomes-separates a modulating method, it is characterized in that, comprises the following step:
Receive a simulated sound input signal and convert a data packet to, and in this data packet, include a plurality of adaptive differential pulse code modulation variable bit data, an initial value and a scale factor, this data packet is a RTP data packet, wherein the certain bits in package gauge outfit is 1, represent that this data packet has an extension field, and this initial value and scale factor system deposit in this extension field;
This data packet is sent to a telecommunication network; And
See through this telecommunication network and receive this data packet, and process according to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this data packet, and then restore a simulated sound output signal.
8. adaptive differential pulse code modulation as claimed in claim 7 becomes-separates modulating method, it is characterized in that, this step that receives this simulated sound input signal comprises:
With one fixedly sampling frequency this simulated sound input signal is sampled and is encoded into the digital sampling value of a plurality of uncompressed;
The digital sampling value of these uncompressed is subtracted each other with an initial value respectively and correspondingly obtain a plurality of digital difference;
According to the difference between the maximum sampling value in the digital sampling value of this initial value and uncompressed in the scheduled time, estimate a scale factor of corresponding this data packet;
These numerical digit difference modulations are become to these adaptive differential pulse code modulation variable bit data; And
These adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this scheduled time are packaged into this data packet.
9. adaptive differential pulse code modulation as claimed in claim 7 becomes-separates modulating method, it is characterized in that, this step that restores this simulated sound output signal comprises:
Understand these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor in this data packet;
Take out these adaptive differential pulse code modulation variable bit data in this data packet;
Take out this initial value and this scale factor in this data packet;
According to these adaptive differential pulse code modulation variable bit data, this initial value and this scale factor, carry out computing, and then obtain the digital sampling value of a plurality of uncompressed; And
The digital sampling value that receives these uncompressed carries out de-quantization processing and restores this simulated sound output signal.
10. adaptive differential pulse code modulation as claimed in claim 7 becomes-separates modulating method, it is characterized in that, this extension field is 32.
11. adaptive differential pulse code modulations as claimed in claim 10 become-separate modulating method, it is characterized in that, this initial value and this scale factor occupy respectively 16 in this extension field.
12. adaptive differential pulse code modulations as claimed in claim 7 become-separate modulating method, it is characterized in that, this telecommunication network is internet, LAN or mobile phone network.
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