Posted at 10.02.2018
A modem to improve communication system performance that uses multiple modulation scheme comprising modulation technique and encoder combinations. As communication system performance and objective change, different modulation schemes may be selected. Modulation schemes may also be selected upon the communication channel scattering function estimate and the modem estimates the channel scattering function from measurements of the channel's frequency (Doppler) and time (multipath) spreading characteristics.
An Adaptive sigma delta modulation and demodulation technique, wherein a quantizer step size is adapted predicated on estimates associated with an input signal to the quantizer, rather than on estimates of input signal to the modulator.
A way of digital conferencing of voice signals in systems using adaptive delta modulation (ADM) with an idle pattern of alternating 1's and 0's has been described. Predicated on majority logic, it permits distortion-free reception of voice of an individual active subscriber by all the other subscribers in the conference. Distortion exists when several subscriber is active and the extent of this distortion depends after the sort of ADM algorithm that is used. An LSI oriented system based on time sharing of an common circuit by lots of channels has been implemented and tested. This system, with only minor changes in circuitry, handles ADM channels that contain idle patterns not the same as alternating single 1's and 0's.
This method used for noise reduction. The modulator factor does not require a big amount of data to be represented. Representation is situated after a frequency domain function having particular characteristics. A preferred embodiment of the invention incorporates transform or sub band filtered signals that are transmitted as a modulated analog representation of an area region of your video signal. The modulation factor reflects this characteristic. Side information specifies the modulation factor
Digital ways to wirelessly communicate voice information. Wireless environments are inherently noisy, so the voice coding scheme chosen for this application must be robust in the presence of bit errors. Pulse Coded Modulation (PCM) and its own derivatives are generally found in wireless consumer products because of their compromise between voice quality and implementation cost. Adaptive Delta Modulation (ADM) is another voice coding scheme, a mature technique that should be considered for these applications due to its bit error robustness and its low implementation cost.
To show the Adaptive Delta Modulation (ADM) voice coding scheme which is the foremost coding scheme procedure when compare to all or any other techniques. The primary part of the procedure is illustrated.
Getting knowledge over different modulation and demodulation techniques
Understanding Delta modulation and Adaptive delta modulation.
Studying Matlab-Simulink which can be used for designing of circuit.
Implementing the circuit in the lab.
Tuning and fixing and calculating its efficiency
Delta modulation is also abbreviated as DM or ‹-modulation. It really is a technique of conversion from an analog-to-digital and digital-to-analog signal. If you want to transmit the voice we utilize this technique. In this technique we do not give that much of importance to the quality of the voice. DM is only the simplest form of differential pulse-code modulation (DPCM). But there is some difference between these two techniques. In DPCM technique the successive samples are encoded into streams of n-bit data. However in delta modulation, the transmitted data is reduced to a 1-bit data stream.
* The analog signal is comparable as a series of segments.
* To find the increase or reduction in relative amplitude, we should compare every single segment of the approximated signal with the initial analog wave.
* By this comparison of original and approximated analog waves we can determine the successive bits for establishing.
* only the change of information is sent, that is, only a rise or decrease of the signal amplitude from the prior sample is sent whereas a no-change condition causes the modulated signal to stay at the same 0 or 1 state of the prior sample.
By using oversampling techniques in delta modulation we can get large high signal-to-noise ratio. Which means the analog signal is sampled at multiple higher than the Nyquist rate.
In delta modulation, it quantizes the difference between the current and the previous step as opposed to the absolute value quantization of the input analog waveform, which is shown in fig 1.
Fig. 1 - Block diagram of an ‹-modulator/demodulator
The quantizer of the delta modulator converts the difference between the input signal and the common of the previous steps. The quantizer is measured with a comparator with reference to 0 (in 2- level quantizer), and its own output is either 1 or 0. 1 means input signal is positive and 0 means negative. Additionally it is called as a bit-quantizer since it quantizes only 1 bit at a time. The output of the demodulator rises or falls because it is nothing but an Integrator circuit. If 1 received means the output raises and if 0 received means output falls. The integrator internally has a low-pass filter it self.
A signum function is accompanied by the delta modulator for the transfer characteristics. It quantizes only degrees of two number and also for at a time only one-bit.
In delta modulation amplitude it is does not matter that there surely is no objection on the amplitude of the signal waveform, due to there exists any fixed amount of levels. In addition to, there is no limitation on the slope of the signal waveform in delta modulation. We can observe whether a slope is overload if so that it can be avoided. However, in transmitted signal there is absolutely no limit to improve. The signal waveform changes gradually.
The interference is due to likelihood of in either DM or PCM is because of limited bandwidth in communication channel. Due to the above mentioned reason 'DM' and 'PCM' operates at same bit-rate.
Another type of DM is Adaptive Delta Modulation (ADM). In which the step-size isn't fixed. The step-size becomes progressively larger when slope overload occurs. When quantization error is increasing with expensive the slope error is also reduced by ADM. By using a low pass filter this should be reduced.
The basic delta modulator was studied in the experiment entitled Delta modulation.
It is implemented by the arrangement shown in block diagram form in Figure
Figure: Basic Delta Modulation
A large step size was required when sampling those elements of the input waveform of steep slope. But a sizable step size worsened the granularity of the sampled signal when the waveform being sampled was changing slowly. A little step size is recommended in regions where the message has a tiny slope.
This suggests the necessity for a controllable step size - the control being sensitive to the slope of the sampled signal. This can be implemented by an arrangement such as is illustrated in Figure
Fig: An Adaptive Delta Modulator
The gain of the amplifier is adjusted in response to a control voltage from the SAMPLER, which signals the onset of slope overload. The step size is proportional to the amplifier gain. This is observed in a youthful experiment. Slope overload is indicated by way of a succession of output pulses of the same sign.
The TIMS SAMPLER monitors the delta modulated signal, and signals when there is no change of polarity over 3 or even more successive samples. The exact ADAPTIVE CONTROL signal is +2 volt under 'normal' conditions, and rises to +4 volt when slope overload is detected.
The gain of the amplifier, and therefore the step size, is made proportional to this Control voltage. Provided the slope overload was only moderate the approximation will 'catch up' with the wave being sampled. The gain will then return to normal before sampler again falls behind.
When coming to comparison of Signal-to-noise ratio DM has larger value than signal-to-noise ratio of PCM. Also for an ADM signal-to-noise ratio in comparison with Signal-to-noise ratio of companded PCM.
Complex coders and decoders are required for powerful PCM. If to increase the resolution we require a sizable quantity of bits per sample. A couple of no memories in Standard PCM systems each sample value is separately encoded into a series of binary digits. An alternative solution, which overcomes some limitations of PCM, is by using past information in the encoding process. Delta modulation is the one way to do to perform source coding.
The signal is first quantized into discrete levels. For quantization process the step size between adjacent samples should be kept constant. In one level with an adjacent one the signal makes a transition of transmission. After the quantization operation is done, sending a zero for a poor transition and a one for a positive transition the signal transmission is achieved. We are able to observe out of this point that the quantized signal must change at each sampling point.
The transmitted bit train would be 111100010111110 for the above mentioned case. The demodulator for a delta-modulated signal is nothing but a staircase generator. To increments the staircase in positively a you need to be received. For negative increments a zero should be receive. That is done by a minimal pass filter in general. The main thing for the delta modulation is to make the right choice of step size and sampling period. A term overloading is occurred when a signal changes randomly fast for the steps to check out. The step size and the sampling period will be the important parameters.
In modern gadgets short-range digital voice transmission is used.
There are numerous products which uses digital techniques. Such as for example cordless telephones, wireless headsets (for mobile and landline telephones), baby monitors are several items. This digital techniques used
Wirelessly communicate voice information. Because of inherent noise in wireless environments the
Voice coding scheme chosen. For this application the existence of robust bit errors must be. Within the presence of bit errors Pulse Coded Modulation (PCM) and its own derivatives are generally used in wireless consumer products. That is due to their compromise between voice quality and implementation cost, but these are not robust schemes.
Another important voice coding scheme is Adaptive Delta Modulation (ADM). It is a mature way of consideration for these kind of applications due to its robustness in bit error and its own low implementation cost.
To quantize the difference between the current sample and the predicted value of the next
Sample ADM can be used. It uses a variable called 'step height' which is used to adjustment of the prediction value of the next sample. For the reproduction of both slowly and rapidly changing input signals faithfully. In ADM, the representation of every sample is one bit (i. e. "1" or "0"). It does not require any data framing for one-bit-per-sample stream to minimizing the workload on the host microcontroller.
In any digital wireless application there must be Bit errors. In ideal environment the majority of the voice coding techniques are given which are good in quality of sound signals. The main thing is to provide good audio tracks signals in everyday environment, there could be a existence of bit errors.
For different voice coding methods and input signals the traditional performance metrics (e. g. SNR) does not measure accurately in music quality.
. "Mean Opinion Score" (MOS) testing is the key important parameter which overcomes the limitations of other metrics by successfully in sound quality. For audio quality the MOS testing can be used. It really is a scale of 1 1 to 5 which tells the music quality status. In there 1 represents very less (bad) speech quality and 5 represents excellent speech quality. A 'toll quality' speech has a MOS score of 4 or more than it. The sound quality of a normal mobile call has same MOS value as above.
The below graphs shows the relationship between MOS scores and bit errors for three of the most common voice coding schemes. Those are CVSD, -law PCM, and ADPCM. A consistently Variable Slope Delta (CVSD) coding is a member of the ADM family in voice coding schemes. The below graph shows the resulted audio tracks quality (i. e. MOS score). All three schemes make clear the amount of bit errors. As the no of bit errors increases the graph indicates that ADM (CVSD) sounds much better than the other schemes which can be can also increase.
In an ADM design error detection and correction typically aren't used because ADM provides poor performance in the existence of bit errors. This contributes to reduction in host processor workload (allowing a low-cost processor to be used).
The superior noise immunity significantly reduced for wireless applications in voice coding method. The ADM is supported strongly by workload for the host processor.
The following example shows the great things about ADM for wireless applications which is demonstrated. For any complete wireless voice product this low-power design is employed which includes all the blocks, small form-factor, including the necessary items.
ADM voice codec
Power supply including rechargeable battery
Microphone, speaker, amplifiers, etc.
Schematics, board layout files, and microcontroller code written in "C".
Delta modulation (DM) may be viewed as a simplified form of DPCM when a two level (1-bit) quantizer is used in conjunction with a set first-order predictor. The block diagram of any DM encoder-decoder is shown below.
The "dm_demo" shows the utilization of Delta Modulation to approximate an input sine wave signal and a speech signal which were sampled at 2 KHz and 44 KHz, respectively. The source code file of the MATLAB code and the out put can be looked at using MATLAB. Notice that the approximated value follows the input value much closer when the sampling rate is higher. You might try this by changing sampling frequency, fs, value for sine wave in "dm_demo" file.
Since DM (Delta Modulator) approximate a waveform Sa (t) by a linear staircase function, the waveform Sa (t) must change slowly in accordance with the sampling rate. This requirement means that waveform Sa (t) must be oversampled, i. e. , at least five times the Nyquist rate.
"Oversampling" means that the signal is sampled faster than is necessary. In the case of Delta Modulation which means that the sampling rate will be much higher than the minimum rate of twice the bandwidth. Delta Modulation requires "oversampling" to be able to obtain an accurate prediction of another input. Since each encoded sample contains a relatively little bit of information Delta Modulation systems require higher sampling rates than PCM systems. At any given sampling rate, two types of distortion, as shown below limit the performance of the DM encoder.
Slope overload distortion: This sort of distortion is due to the use of your step size delta that is too small to check out portions of the waveform that contain a steep slope. It can be reduced by increasing the step size.
Granular noise: This results from utilizing a step size that is too large too large in elements of the waveform having a tiny slope. Granular noise can be reduced by decreasing the step size.
Even for an optimized step size, the performance of the DM encoder may still be less satisfactory. Another solution is to employ a variable step size that adapts itself to the short-term characteristics of the source signal. This is the step size is increased when the waveform has a step slope and decreased when the waveform has a relatively small slope. This strategy is called adaptive DM (ADM).
While transmitting speech for e. g. telephony the transfer rate should be kept no more than possible to save lots of bandwidth because of monetary reason. For this function Delta Modulation, adaptive Delta modulation, Differential Pulse-Code modulation is utilized to compress the info.
In this different kind of Delta modulations and Differential Pulse Code modulations (DPCM) were realized to compress music data.
At first the principal of compressing music data are explained, that your modulations predicated on. Mathematical equations (e. g. Auto Correlation) and algorithm (LD recursion) are used to develop solutions. Based on the mathematics and principals Simulink models were implemented for the Delta modulation, Adaptive Delta modulation as well as for the adaptive Differential Pulse Code modulation. The theories were verified through the use of measured signals on these models.
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals, which was invented by Alec Reeves in 1937. It is the standard form for digital music in computers and different COMPACT DISK and DVD formats, as well as other uses such as digital telephone systems. A PCM stream is an electronic representation of your analog signal, where the magnitude of the analogue signal is sampled regularly at uniform intervals, with each sample being quantized to the nearest value within a variety of digital steps.
PCM streams have two basic properties that determine their fidelity to the original analog signal: the sampling rate, which is the amount of times per second that samples are taken; and the bit-depth, which determines the number of possible digital values that every sample may take.
In conventional PCM, the analog signal may be processed (e. g. by amplitude compression) before being digitized. After the signal is digitized, the PCM signal is usually put through further processing (e. g. digital data compression).
PCM with linear quantization is known as Linear PCM (LPCM).
Some kinds of PCM incorporate signal processing with coding. Older versions of the systems applied the processing in the analog domain as part of the A/D process; newer implementations do it in the digital domain. These simple techniques have been largely rendered obsolete by modern transform-based sound compression techniques.
* DPCM encodes the PCM values as variations between your current and the predicted value. An algorithm predicts another sample predicated on the previous samples, and the encoder stores only the difference between this prediction and the actual value. In case the prediction is reasonable, fewer bits may be used to represent the same information. For audio, this kind of encoding reduces the number of bits required per sample by about 25% in comparison to PCM.
* Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the quantization step, to permit further reduced amount of the mandatory bandwidth for confirmed signal-to-noise ratio.
* Delta modulation is a form of DPCM which uses one bit per sample.
In telephony, a typical music signal for an individual phone call is encoded as 8000 analog samples per second, of 8 bits each, giving a 64 Kbit/s digital signal known as DS0. The default signal compression encoding on a DS0 is either -law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and almost all of all of those other world). They are logarithmic compression systems in which a 12 or 13-bit linear PCM sample number is mapped into an 8-bit value. This technique is described by international standard G. 711. An alternative proposal for a floating point representation, with 5-bit mantissa and 3-bit radix, was abandoned.
Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal even further. An ADPCM algorithm can be used to map some 8-bit -law or A-law PCM samples into some 4-bit ADPCM samples. In this manner, the capacity of the line is doubled. The technique is detailed in the G. 726 standard.
Later it was discovered that even further compression was possible and additional standards were published.
Pulse code modulation (PCM) data are transmitted as a serial bit blast of binary-coded time-division multiplexed words. When PCM is transmitted, premodulation filtering shall be used to confine the radiated RF spectrum. These standards define pulse train structure and system design characteristics for the implementation of PCM telemetry formats.
The PCM formats are divided into two classes for reference. Serial bit stream characteristics are described below prior to frame and word oriented definitions.
Two classes of PCM formats are covered in this chapter: the essential, simpler types are class I, and the more technical applications are class II. The usage of any class II technique requires concurrence of the range involved. All formats with characteristics described in these standards are class I except those determined as class II. The following are types of class II characteristics:
a. Bit rates higher than 10 megabits per second
b. Word length in excess of 32 bits.
c. fragmented words
d. more than 8192 bits or 1024 words per minor frame.
e. uneven spacing, not within the definition of sub commutation or super commutation
f. Format changes.
g. asynchronous embedded formats
h. tagged data formats.
i. packet telemetry
j. formats with data content apart from unsigned straight binary, discrete, or complement arithmetic representation for negative numbers such as floating point variables, binary-coded decimal, and gain-and-value
k. asynchronous data transmission
l. merger of multiple format types
There are several means of demodulation depending how parameters of the base-band signal are transmitted in the carrier signal, such as amplitude, frequency or phase. For instance, for a sign modulated with a linear modulation, like AM (Amplitude Modulated), we may use a synchronous detector. Alternatively, for a sign modulated with an angular modulation, we should use an FM (Frequency Modulation) demodulator or a PM (Phase Modulation) demodulator. Different sorts of circuits perform these functions.
Many techniques-such as carrier recovery, clock recovery, bit slip, frame synchronization, rake receiver, pulse compression, Received Signal Strength Indication, error detection and correction, etc. -- are just performed by demodulators, although any specific demodulator may perform only some or none of the techniques.
One important attribute of demodulation (or demod) data is the fact that it focuses on high frequency vibration. Utilizing a high pass filter, low frequency data is filtered out and a data collector can "zoom in" on low level high frequency vibration. Which means that some peaks that could otherwise be lost in the noise floor of a normal narrow band spectrum (lower than the normal vibration a machine emits) can
be detected using demodulation techniques.
Another feature of demod, or of high frequency vibration in general, is that it is easily attenuated and will not travel well via a machine's structure (termed the "disco effect"). As one moves away from a loud music source, one tends to hear only the bass, or low frequency sound, since the treble or high frequency sounds dissipate rather quickly. Therefore that vibration detected with demod is usually produced locally. Regarding a motor driving a pump through a coupling, demod data collected on the pump end will usually reflect the vibration emitted by the pump end. Lower frequency vibration may be transmitted through the coupling and may even be amplified on the other end of the device depending upon its mobility.
Short-range wireless digital voice transmission is employed extensively in modern-day consumer electronics. Products such as cordless telephones, wireless headsets (for mobile and landline telephones) and baby monitors are just a several items that use digital techniques to wirelessly communicate voice information.
Wireless environments are inherently noisy, so the voice coding scheme chosen for such an application must be robust in the occurrence of bit errors.
Pulse coded modulation (PCM) and its derivatives are generally found in wireless consumer products for their compromise between voice quality and implementation cost, but these schemes are not particularly robust in the occurrence of bit errors. Adaptive delta modulation (ADM) is an adult technique that needs to be considered for these applications due to its bit error robustness and its own low implementation cost.
ADM is a voice coding technique that quantizes the difference between your current sample and the predicted value of the next sample. It uses a variable 'step height' to adapt the predicted value of another sample so that both slowly and rapidly changing input signals can be faithfully reproduced. One bit is employed to represent each sample in ADM. The one-bit-per-sample ADM data stream requires no data framing, thereby minimizing the workload on the host microcontroller.