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A first note: Most modern text-to-speech systems, like the one from AT&T you have linked to, use concatenative speech synthesis. This technique uses a large database of recordings of one person's voice uttering a long collection of sentences - selected so that the largest number of phoneme combinations are present. Synthesizing a sentence can be done ...

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Most certainly not. While there has been some claims to break Shannon here and there, it usually turned out that the Shannon theorem was just applied in the wrong way. I've yet to see any such claim to actually prove true. There are some methods known that allow for transmission of multiple data streams at the same time on the same frequency. The MIMO ...

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For many years the state of the art was to use a convolutional "inner code" and a block "outer code". The "inner" and "outer" terminology come from the following block diagram: $$\boxed{\scriptstyle \rm Payload}{\longrightarrow}\boxed{\scriptstyle\textrm{Outer Encode}}{\longrightarrow}\boxed{\scriptstyle\textrm{Inner Encode}}{\longrightarrow}\boxed{\... 14 The capacity of a channel should be viewed as analogous to the speed limit on a highway. It is possible to travel at a speed greater than the posted limit on a highway but it is not possible to achieve good gas mileage while doing so. Similarly, it is possible to transmit data at rates higher than the capacity of the channel (in fact, unlike highways, ... 11 suppose, Carrier signal frequency = 2800KHz message signal frequency = 3KHz Then you will get a signal that looks like this in the frequency plane. Obviously this is not to scale, but you get the idea. but what will happen if it is reversed ? i.e Message signal frequency = 2800KHz Carrie signal frequency = 3KHz please explain would happen here ? Then ... 10 You need to integrate the modulating signal because frequency is the time derivative of phase. Therefore, the typical relationship from introductory calculus holds:$$ \phi_i(t) = \int_{-\infty}^{t} \frac{d\phi_i(\tau)}{d\tau} d\tau = \int_{-\infty}^{t} 2\pi f_i(\tau) d\tau $$For causal signals, the lower limit on the integral changes to zero:$$ \phi_i(...

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The AM / FM category is a bit strange for effects - true, you can get some interesting effects by modulating the amplitude of an input signal ; but what does "frequency modulation" mean for a complex input signal for which you don't even have an accurate frequency representation? You could very well say that every effect is an amplitude modulation, by the $\... 8 Ideal symbol pulses are sequences of rectangular pulses. Rectangular pulses, unfortunately, have horrible frequency profiles, as shown below. The main lobe is where most of the pulse power is, and the other lobes are called side lobes. Though they do diminish in power the farther out they go, they don't diminish very quickly. The main purpose of pulse ... 8 Direct-sequence spread spectrum (DSSS) is a technique that is used to generate a modulated signal that occupies more bandwidth than would be implied by its information content alone. A DSSS signal is generated via multiplication of a (typically digitally-modulated) baseband signal by another spreading code waveform. The spreading code waveform is constructed ... 8 Carrying information requires bandwidth. For a given S/N ratio, modulating a signal to carry more information will thus expand its bandwidth. Call the addition bandwidth "side bands". If you don't add side bands to a fixed frequency carrier, you can't expand its bandwidth, and thus you can't transmit any information (other than the presence of a ... 8 The filter that you're referring to is called a preselection filter. Its purpose is to filter out everything but the desired signal of interest before mixing to baseband. Unwanted components could include other signals that are nearby in frequency, or just noise that lies outside the desired signal's bandwidth. Preselection can serve multiple purposes: It ... 8 All implementation aspects aside, the constellation you propose performs worse than QPSK in an additve white gaussian noise (AWGN) channel. I claim this based on simulations that I have run with Matlab calculating the symbol error rate (SER) as a function of signal-to-noise ratio (SNR). Here is the result: As you can see, for a given SNR, the proposed ... 8 I am currently implementing acoustic FSK modulation and demodulation. I am not a signal processing guy… Since you say you have matched filters, and you mention non-coherent detection, I think you're pretty much of a digital communication person already – the step to being a DSP person is pretty small :) The fully-fledged synchronizer SDR approach So, the ... 8 The NCO is a cyclical counter that can go on indefinitely but is otherwise similar to what you suggest in that you are increment n to set the output rate. It basically is a look up table of all the values in one complete cycle, and "wraps" on an overflow so that it will output continuous cycles with no discontinuity. I think the NCO is ideal for what you ... 8 As you have realized, the hard part of doing digital communications is carrier, symbol and frame synchronization, and channel estimation/equalization. The bad news is that you can't get around these problems. The good news is that implementing these is not that hard, as long as you limit yourself to narrowband BPSK. I know, because I have done this myself, ... 7 There is no inherent symbol rate for any modulation type. 64 QAM symbols transmit 6 bits per symbol, and 256 QAM symbols transmit 8 bits per symbol. If the bit rate were to stay constant, then the 256 QAM symbol rate would be 6/8 of the 64 QAM symbol rate, but it appears that that is not the case (assuming your numbers are correct). That may be because ... 7 As said by Jason, this might simply be that you are not implementing your oscillator correctly - for example by multiplying the frequency by time instead of integrating it. Note also - and this is unrelated to your topic but really worth observing - that your formula for frequency modulation implements a behavior very different to that of most synthesizers, ... 7 The$BT$product is the bandwidth-symbol time product where$B$is the$-3\textrm{ dB}$(half-power) bandwidth of the pulse/filter and$T$is the symbol duration. For different applications you will find varying recommended values. In GSM telephony for instance, a$BT=0.3$is recommended. In satellite communications with GMSK, for near-earth missions the ... 6 Yes and no. The mapping is arbitrary as long as the receiver correctly determines which constellation point a symbol is. If the receiver makes a mistake, though, it is most likely going to pick a "neighbor" constellation point (i.e. a constellation point that is only one spot away). It is highly unlikely that a correctly implemented receiver will pick a ... 6 This is actually a really tough problem because of the channel characteristics. Most computer speakers have fairly limited bandwidth, have significant non-linearities and the room acoustics are often time variant. Life becomes A LOT easier if you can just run a cable from the headphone output of one PC into the line input of the other. 6 The sinusoid test signal is given by $$s(t) = V_0\cos(2\pi f_0)$$ and its Fourier transform by $$S(f) = \frac{V_0}{2}\big[\delta(f-f_0) + \delta(f+f_0)\big].$$ Inserted in$S_\mathrm{AM-DSB-C}$it yields $$S_\mathrm{AM-DSB-C}(f) = \frac{AV_0}{4}\big[\delta(f-f_c-f_0) + \delta(f-f_c+f_0) + \delta(f+f_c-f_0) + \delta(f+f_c+f_0) \big] + \frac {Ac}{2} \big[\... 6 You need to build a time varying delay, where you can modulate the delay amount over time. The peak delay modulation is a function of your maximum desired frequency shift and the modulation frequency. This is not trivial since it will require fractional sample delays with some kind of interpolation algorithm. You can't round to the nearest integer delay ... 6 What is the advantage of performing the FSK using IQ modulation? You only need one RF oscillator operating at a single frequency, instead of having 2 (or more in the case of M-ary FSK) oscillators operating at separate frequencies for each bit/symbol. Since you only have one oscillator, you don't have to worry about discontinuities in the phase of the ... 6 The SDR (or any general digital signal processing system) takes the received RF signal, and downconverts it from the carrier frequency to the baseband. Now, the real bandpass signal from the antenna does not necessarily have a symmetric spectrum around the carrier frequency, but it can be arbitrary. If the downconverter now shifts the spectrum to the center ... 6 The optimal decision regions are the Voronoi Regions. I dont know, if this is what you are looking after. import numpy as np points = np.array([(1,1), (1,-1), (-1,1), (-1,-1), (3,3), (3,0), (3,-3), (0,-3), (-3,-3), (-3,0), (-3,3), (0,3), (5,0), (0,5), (-5,0), (0,-5)]) from scipy.spatial import Voronoi, voronoi_plot_2d vor = Voronoi(points) voronoi_plot_2d(... 6 As explained in Maximilian Matthé's answer, the exact computation of the symbol error probability of this constellation (ITU-T V.29 modem standard) is quite complex. However, you can quite easily compute an approximation which becomes very good for relatively large signal to noise ratios (SNRs). This approximation is based on the union bound. The symbol ... 6 There are a couple reasons. One is that (1-z^{-1}) represents x[n]-x[n-1] which is a finite difference over a very small period of time. and that is an approximation to a differentiator. The reciprocal is$$ \frac{1}{1-z^{-1}} = \frac{z}{z-1}$$which is the inverse operator. We normally call the inverse operation of differentiation, we call that "... 6 1) Is there a connection between the modulation kind and the channel capacity? Channel capacity is usually defined as the number of information (usually measured by the number of bits) can be sent per channel use to get arbitrarily low number of errors, but we don't know exactly how (random coding). A channel use can be thought as a modulation symbol ... 5 I interpret the question as follows: If we modulate a carrier with a pure tone using AM, we get a single set of sidebands, but if we modulate with phase modulation, we get an infinite number of sidebands, spaced at the modulation frequency. Why? It is easy to see why amplitude modulation at a single frequency gives exactly two sidebands. Simply multiply ... 5 Two successive symbols in the demodulator are$Z_1 = (X_1,Y_1)$and$Z_2 =(X_2,Y_2)$where$X$is the output of the I branch and$Y$the output of the Q branch of the receiver. The hard-decision DBPSK decision device considers the question: Is the new symbol$Z_2$closer to the old symbol$Z_1$or to the negative$-Z_1\$ of the old symbol? and thus ...

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