# Weaver SSB Modulation/Demodulation - A Tutorial

Save this PDF as:

Size: px
Start display at page:

## Transcription

1 Weaver SSB odulation/demodulation - A Tutorial Derek Rowell February 18, Introduction In 1956 D. K. Weaver 1 proposed a new modulation scheme for single-sideband-suppressedcarrier (SSB) generation. The Weaver ethod (also known as the Third ethod), has potential advantages compared to the two prior methods, the filter and phase-shift methods, but is more difficult to understand. With the advent of DSP (Digital Signal Processing) and SDR (Software-Defined Radio) the Weaver ethod has received much greater attention but is still not generally well understood by hobbyists. This tutorial is designed to introduce the Weaver ethod using simple mathematics that is limited to a few trigonometric identities. The descriptions described here are based on real (as opposed to complex) signal representations. In particular all signals are represented in terms of sinusoidal components of the form A cos(2πf t + φ) where A is the amplitude, F is the frequency with units of Hz, and φ is the phase (with units of radians) (0 φ < 2π). For convenience we will make the substitution ω = 2πF and express the frequency as an angular frequency, with units radians/second. Fourier theory tells us that any practical time-based waveform, f(t), may be expressed mathematically as a sum (finite or infinite) of such sinusoids, f(t) = A n cos(ω n t + φ n ), (1) and can be entirely described by its spectrum, that is the collection of all amplitudes A n, frequencies ω n, and phases φ n 2. We are all familiar with this concept in terms of filtering a signal, where we modify that signal by manipulating its spectrum, that is modifying the values of A n and φ n. For the purposes of this tutorial we need to be able to think of a signal in terms of its spectral components. 1 Weaver, D.K., A third method of generation and detection of single sideband signals, Proc. IRE, Dec. 1956, pp There are several ways of expressing the spectrum of a signal f(t). In this tutorial we adopt the convention of a one-sided, purely real spectrum as described above. Another more general and compact description uses a collection of complex exponentials of the form A n e ±jωnt to describe the spectrum, and requires positive and negative frequency components. odern signal analysis relies almost exclusively on the complex notation. 1

2 1.1 Introduction to SSB Communication Single-sideband suppressed carrier (SSB) modulation is a variant of amplitude modulation (A) and is used extensively in high-frequency radio voice communication, such as maritime and ham radio voice communication. Amplitude (A) odulation: In A radio, the audio information is impressed upon a sinusoidal carrier signal cos(ω c t) with angular frequency ω c, and amplitude A by modulating the amplitude of the carrier A cos(ω c t), with an audio signal f audio (t) as follows: f A (t) = (1 + αf audio (t)) cos(ω c t) (2) where α is known as the modulation index, and must be chosen so that αf audio (t)) > 0 (Otherwise distortion caused by over modulation occurs.). Notice if the audio is silent that is f audio (t) = 0, the transmitted signal is simply the carrier f A (t) = A cos(ω c t). For simplicity we start by assuming we have a very simple audio signal, a simple audio tone with a frequency of F a Hz, so that f audio (t) = A a cos(2πf a t + φ a ) = A a cos(ω a t + φ a ) (3) where A a is the amplitude, ω a = 2πF a is the angular frequency in units of radians/second, and φ a is an arbitrary phase angle 3 When Eq (3) is substituted into Eq (2) f A (t) = cos(ω c t) + αa a (cos(ω a t + φ a ) cos(ω c t)) (4) where there is a multiplication of two time-based sinusoidal functions. Using Eq. (A.1) in the appendix to expand the product of two cosine functions) f A (t) = cos(ω c t) + A aα 2 cos((ω c + ω a )t + φ a ) + A aα 2 cos((ω c ω a )t φ a ) (5) which shows three separate sinusoidal components 1) a carrier component at frequency ω 2, 2) an upper sideband component at frequency (ω c + ω a ), and 3) a lower sideband component at frequency (ω c ω a ), as shown in Fig. 1. As noted above, we can represent any audio signal, for example speech or music, as a sum of N such components and write f audio (t) = A n cos(ω n t + φ n ), and substitute into Eq (2). Then applying the above argument to each of the N components ( f A (t) = cos(ω c t) + α N ) ( A n cos((ω c + ω n )t + φ n ) + α N ) A n cos((ω c ω n )t φ n ) 2 2 (6) where the upper and lower sidebands each contain many components, as shown in Fig We will use angular frequency throughout this document because it generates a more compact representation. 2

3 )? = = F E J A = E ) = F E J A? = H H E A H A H I A > ) H B I E C = K F F A H I A > = ) =? =?? Figure 1: Amplitude spectrum of a simple A rf signal with carrier frequency ω c and audio frequency ω a. = F E J A = F E J A = E I E C = ) H B I E C =? = H H E A H ) A H I A > K F F A H I A > = N? = N Figure 2: Amplitude spectrum of an A rf signal with carrier frequency ω c modulated by an audio signal containing a set of three components with frequencies 0 < ω < ω max. The envelope of the audio spectrum is shown as shaded, and asymmetric to help visualize the spectral relationships between the sidebands. We are all familiar with A modulation though listening on the A broadcast band. Historically, A was the first radio voice/music communication method, and while easy to implement in hardware it, has a number of disadvantages compared to other modulation techniques. In particular, for high frequency communication ost of the transmitted signal power is contained within the carrier (which is transmitted continuously, even in the absence of an audio signal), making A a very energy inefficient communication mode. The spectral width of the A waveform is twice the audio bandwidth, and is therefore an inefficient use of the high-frequency radio spectrum. In long-distance communication, with multipath reflections from the ionosphere, cancelation of the carrier can occur leading to selective fading, and loss of intelligibility. Single-Sideband Suppressed Carrier (SSB) odulation: SSB generation simply involves elimination of the carrier and one of the two sidebands from an A transmission, 3

4 ?? leaving only the upper (USB) or lower (LSB) sideband. There are therefore two possible modes of SSB transmission, either LSB or USB, as shown in Fig * = F E J A 7 5 * H B I E C = K F F A H I A > = F E J A = E? = N = N 5 * = F E J A A H I A > 5 * H B I E C =? = N Figure 3: Spectral representation of upper-side-band (USB), and lower-sideband (LSB) signals of a three component audio signal with audio bandwidth ω max at the same rf frequency ω c. Notice that SSB transmission directly addresses the first two criticisms of A transmission noted above: There is no power transmitted as a carrier, and the power transmitted is zero in the absence of any audio signal. The spectral bandwidth is half that of an A signal, In practice SSB is also far less susceptible to the selective fading that plagues A transmission at high frequencies. We note that USB modulation is a straight frequency translation of each component in f audio (t) : f audio (t) = A n cos(ω n t + φ n ) USB A n cos((ω c + ω n )t + φ n ), (7) and LSB is a translation and reflection (flipping) of the components about the suppressedcarrier frequency ω c : f audio (t) = A n cos(ω n t + φ n ) LSB A n cos((ω c ω n )t φ n ). (8) 4

5 1.2 Historical methods of SSB Waveform Generation: Historically there were two methods of generating an SSB signals: The filtering method, is a brute-force method, in which a multiplier is used to create a double sideband (DSB) waveform at a fixed frequency ω c. If f audio (t) = A a cos(ω a t + φ a ) f DSB (t) = A a cos(ω a t + φ a ) cos(ω c t) = A a 2 cos((ω c + ω a )t + φ a ) + A a 2 cos((ω c ω a )t φ a ) from Eq. (A.1). f DSB (t) is then passed through a very high quality band-pass filter (in hardware usually a mechanical or xtal filter) centered on the desired sideband to reject the unwanted sideband. The phase-shift or phasing method generates the SSB signal directly from the trigonometric relationships in Eq. (A.5) in the Appendix cos(a + b) = cos(a) cos(b) sin(a) sin(b) cos(a b) = cos(a) cos(b) + sin(a) sin(b) with the constant angles a, b replaced with time varying functions of the form ωt. If the audio signal is a simple tone f audio (t) = A a cos(ω a t + φ a ), a USB signal may be created from the expansion y USB (t) = A a cos((ω c +ω a )t+φ a ) = A a cos(ω a t+φ a ) cos(ω c t) A a sin(ω a t+φ a ) sin(ω c t). There is a problem because we do not have the signal A a sin(ω a t + φ a ) available to us, and in practice it cannot be generated directly from the audio A a cos(ω a t + φ a ). However, using Eq. (A.10) in the Appendix (sin(a) = cos(a π/2)) we can write y USB (t) = A a cos(ω a t + φ a ) cos(ω c t) A a cos(ω a t + φ a π/2) sin(ω c t). (9) and use a (hardware or software) all-pass π/2 phase-shifter (also known as a Hilbert transformer) to approximate the A a sin(ω a t + φ a ) as indicated in Fig. (4). ' F D = I A I D E B J A H? I J? I J F I E J 0 E > A H J J H = I B H A H Figure 4: The Hilbert transformer for approximating a wideband π/2 phase shift. Similarly for lower-sideband generation 4 y LSB (t) = A cos((ω c ω a )t + φ) = A cos(ω a t + φ) cos(ω c t) + A sin(ω a t + π/2) sin(ω c t) = A cos(ω a t + φ) cos(ω c t) + A cos(ω a t + φ π/2) sin(ω c t) (10) 5

6 = )? I J = = = ' F D = I A I D E B J A H 0 E > A H J J H = I B H A H? I J? = E E F K J B H A G K A? O G K H = J K H A? I? E = J H B J ) =? I = J = ) I E J = = = I E J? 5 5 * K J F K J O J B H 7 5 * B H 5 * Figure 5: The signal processing steps involved in the phasing method of SSB generation.note that the only difference between USB and LSB generation is the sign of the summation at the output. The signal processing flow is indicated in Fig. 5. The phase-shift method is probably the most widely used SSB generator, and is well suited to both hardware and software (DSP) methods. Its major downside is that the phase shifter must be an all-pass filter, with a constant phase shift of π/2 radians (90 ) across the whole audio bandwidth. Although modern filter design allows for accurate results, this can at best be only approximated. Errors in the phase shift result in incomplete suppression of the unwanted sideband. 2 The Weaver SSB odulator In the 1950 s the design and implementation of accurate phase- shift networks was a difficult task. The Weaver method eliminates the need for such networks, and replaces them with a pair of identical low-pass filters with modest design specifications. 2.1 The Weaver odulator As discussed above, our goal is to translate an audio signal f audio (t) containing N spectral components f audio (t) = A n cos(ω n t + φ n )) to a SSB waveform y ssb = A i cos((ω c ± ω n )t ± φ n ) where the + indicates the USB, and the the LSB modes. 4 The explicit use of the multiplication symbol in Eqs. (6) and (7) has no mathematical significance, and is only used to emphasize the signal processing operations implied. 6

7 Assume that the audio signal is band limited between frequencies ω min and ω max as shown in Fig. 6. For example, in a communications radio the audio might be limited to Hz (1, , 849 rad/s). Then define the mid-point of the audio spectrum as ω o = (ω min + ω max )/2. (11) = F E J A = E I E C = C K = > J D E E E = N E = N = N Figure 6: Schematic representation of the amplitude spectrum of the band-limited audio signal f audio (t). The shaded area shows the envelope of the A n. Also shown is the definition of the frequency ω o = (ω min + ω max )/2. The block diagram of the Weaver modulator is shown in Fig. 7. Notice that there are a pair of parallel chains, each containing two oscillators, a low-pass filter, and two multipliers. As before, for simplicity we will consider the processing of a single representative component of f audio (t), f a (t) = A a cos(ω a t + φ a )) with amplitude A a, frequency ω a, and phase φ a through the modulation process. The first oscillator in each chain is a fixed, low frequency oscillator set to the audio mid-point frequency ω o. The second is a high frequency (rf) oscillator at a frequency ω s. In practice this is used for tuning the SSB output to the desired frequency ω c. (As we will see below ω s ω c ). The two low-pass filters are identical, and have a cut-off frequency of ω o. For now they are assumed to be ideal - that is they pass all components with a frequency ω o ω ω o unimpeded, while rejecting all frequencies. We now examine the flow of f a (t) through the modulator: 1. At the output of the first multipliers (f 1 (t) and f 2 (t)): The output of the first pair of multipliers are the products of the audio signal f a (t) = A a cos(ω a t + φ a ) and the pair of quadrature sinusoids cos(ω o t) and sin(ω o t): f 1 (t) = f a (t) cos (ω o t) = A a cos (ωt + φ a ) cos (ω o t) = A a 2 cos ((ω a ω o ) t + φ a ) + A a 2 cos ((ω a + ω o ) t + φ a ) (12) 7

8 5 B J F = I I B E J A H B! J B # J = E I E C = B J = E? I J I E J B J B E N L = H E = > A F = I I B E J A H I? I I J I E I J B J " B J \$ 5 5 * K J F K J O J Figure 7: The signal processing flow for the Weaver SSB modulator from Eq. (A.1), and f 2 (t) = f a (t) sin (ω o t) = A a cos (ω a t + φ a ) sin (ω o t) = A a 2 sin ((ω a ω o ) t + φ a ) + A a 2 sin ((ω a + ω o ) t + φ a ) (13) from Eq. (A.6). The amplitude spectra of f 1 (t) and f 2 (t) are shown in Fig. 8. = F E J A = F E J A = E I E C = B J B J E E J E = K J E F E? = J E E = N = N E E = N = N Figure 8: Schematic amplitude spectra at the output of the first multiplier stage in the Weaver modulator. Two sample audio components are shown as dashed lines. Note that although f 1 (t) and f 2 (t) have identical amplitude spectra, their phase spectra differ. 2. At the outputs of the Low-Pass Filters (f 3 (t) and f 4 (t)): We assume that the two low-pass filters are ideal, that is they pass all spectral components(positive or negative) within their bandwidth ω o ω ω o unimpeded, while completely rejecting all components with ω < ω o and ω > ω o, where the response to negative frequency 8

9 components is as defined in Appendix B. 5 The outputs from the two low-pass filters, (f 3 (t) and f 4 (t)), are therefore simply the low-frequency (difference) components in f 1 (t) and f 2 (t) with frequencies below ω o : and f 3 (t) = f 4 (t) = { Aa 2 cos ((ω a ω o ) t + φ a ) if ω o ω a ω o 0 otherwise. { A a 2 sin ((ω a ω o ) t + φ a ) if ω o ω a ω o 0 otherwise. The amplitude spectra of f 3 (t) and f 4 (t) are shown in Fig. 9. (14) (15) = F E J A = F E J A B J B J B J B J! " F = I I B E J A H? K J B B B H A G K A? O = N E E = N Figure 9: Low-pass filtering to retain only those components with a frequency in the range ω o ω a ω o. 3. At the outputs of the second multipliers (f 5 (t) and f 6 (t)): In general the second oscillator frequency ω s >> ω o. Then f 5 (t) and f 6 (t) are simply: f 5 (t) = f 3 (t) cos(ω s t) = A a 4 cos (((ω s + ω o ) ω a ) t φ a ) + A a 4 cos (((ω s ω o ) + ω a ) t + φ a ) (16) f 6 (t) = f 4 (t) sin(ω s t) = A a 4 cos (((ω s + ω o ) ω a ) t φ a ) + A a 4 cos (((ω s ω o ) + ω a ) t + φ a )(17) 4. Summation/Subtraction to Generate the SSB Output: Upper or lower sideband signals are generated by simple addition or subtraction of f 5 (t) and f 6 (t): y usb (t) = f 5 (t) + f 6 (t) = A a 2 cos (((ω s ω o ) + ω a ) t + φ a ) = A a 2 cos ((ω c + ω a ) t + φ a ) (18) 5 For theoretical reasons it is impossible to design or build an ideal filter. We simply assume its existence to illustrate the the signal processing involved and accept that any practical filter will not meet the idealized specifications. 9

10 where ω c = ω s ω o, which is the desired signal frequency. similarly y lsb (t) = f 5 (t) f 6 (t) = A a 4 cos (((ω s + ω o ) ω a ) t φ a ) = = A 2 cos ((ω c ω a ) t φ a ) (19) where in this case ω c = (ω s + ω o ) is the output LSB frequency. 2.2 Design Summary: Given a band-limited audio signal f audio (t) = A n cos(ω n t + φ n ) where ω min ω n < ω max for all n, proceed as follows: and 1. Select ω o = (ω min + ω max /2. 2. Design and implement a pair of low-pass filters with cut-off frequency ω o, (and with a transition band of width 2ω min ). To generate a USB signal at frequency ω c : y usb (t) = A n cos(ω c + ω n )t + φ n ), Set the second oscillator frequency to ω s = ω c + ω o Set the output summer so that y usb (t) = f 5 (t) + f 6 (t). To generate a LSB signal at frequency ω c : y lsb (t) = A n cos(ω c ω n )t φ n ), Set the second oscillator frequency to ω s = ω c ω o Set the output summer so that y lsb (t) = f 5 (t) f 6 (t). 3 Weaver Demodulation While the modulator has a well defined task of translating a well defined low frequency, band-limited signal to a high frequency rf signal, the demodulation process has a particularly 10

11 5 messy working environment in the sense that its input is potentially the whole rf spectrum, including the A, F, TV, etc bands. From the cacophony of signals it must select a narrowband signal and demodulate it without interference from adjacent (and distant) signals. The Weaver demodulator does an excellent job of this, and is basically a complete SSB communications receiver on its own. As shown in Fig. 10 the Weaver demodulator is basically the reverse of the steps used in the modulator. H B I E C = B J F = I I B E J A H B! J B # J B J H B? I I J I E I J B J I L = H E = > A B E N F = I I B E J A H? I J I E J B J " B J \$ = E K J F K J O J Figure 10: The structure of the Weaver Demodulator. We start with a clean high frequency rf signal f rf (t) = A a cos(ω a t+φ a ).Notice that the major difference from the Weaver modulator is the reversal of the order of the high frequency and low frequency oscillators. The first step is to multiply the input waveform f rf (t) by a pair of variable frequency oscillators (cos(ω s t and sin(ω s t)) to translate the input down to baseband. 1. At the output of the first multipliers (f 1 (t) and f 2 (t)): Assume that the input is a broadband RF signal with bandwidth ω max as shown in Fig. 11, with a narrowband signal centered on ω s. The first step is to multiply f rf (t) by the quadrature pair cos(ω s t) and sin(ω s t): f 1 (t) = f rf (t) cos(ω s t) = A cos(ω a t + φ a ) cos(ω s t) = A 2 cos ((ω a ω s )t + φ a ) + A 2 cos ((ω a + ω s ) + φ a ) (20) f 2 (t) = f rf (t) sin(ω s t) = A cos(ω a t + φ a ) sin(ω s )t = A 2 sin ((ω a ω s )t + φ a ) + A 2 sin ((ω a + ω s )t + φ a ) (21) The effect of this step is to translate the desired signal down to baseband, as shown in Fig. 11. Note the doubling of the bandwidth in f 1 (t) and f 2 (t), this will become important when we discuss aliasing in DSP implementations later. 11

12 I I = F E J A 4. I E C = B J H B = F E J A B J B J B E H I J K J E F E A H I = N = I = N Figure 11: The first step in Weaver demodulation: the broadband rf input signal f rf (t) is multiplied by the quadrature pair cos(ω s t and sin(ω s t). The line represents a component with frequency ω a in a signal shown as the dark shaded area. 2. Low pass filter outputs: Assume the low-pass filters have a cut-off frequency of ω o and will pass components in the frequency range ω o ω ω o, (see Appendix B). Assume ω s >> ω o and ω c >> ω o. Clearly the sum terms in Eqs. (20) and (21) and will be rejected by the filter, and the baseband components retained only if they fall within the passband of the filter. f 3 (t) = { A 2 cos ((ω a ω s )t + φ a ) if ω 0 ω a ω s ω o, 0 otherwise, (22) and f 4 (t) = { A 2 sin ((ω a ω s )t + φ a ) if ω o ω a ω s ω o, 0 otherwise. (23) as shown in Fig. 12. = F E J A B J B J = F E J A B J B J! " F = I I B E J A H I = N Figure 12: The second step in Weaver demodulation: the action of the low-pass filters to eliminate all components outside the filter passband ( ω o ω a ω o ). 3. Second multiplier outputs: f 5 (t) = f 3 (t) cos(ω o t) = A 2 cos ((ω a ω s )t + φ a ) cos(ω o t) = A 4 cos ((ω a (ω s + ω o )) t + φ a ) + A 4 cos ((ω a (ω s ω o )) t + φ a ) (24) 12

13 only if ω o ω a ω s ω o, and f 6 (t) = f 4 (t) sin(ω o t) = A 2 sin ((ω a ω s )t + φ a ) sin(ω o t) = A 4 cos ((ω a (ω s ω o )) t + φ a ) + A 4 cos ((ω a (ω s + ω o )) t + φ a ) (25) also only if ω o ω a ω s ω o. 4. Summer output: The demodulated audio waveform y audio (t) is found at the output of the summer: y (+) audio (t) = f 5(t) + f 6 (t) = A 2 cos((ω a (ω s ω o ))t + φ a ) (26) y ( ) audio (t) = f 5(t) f 6 (t) = A 2 cos((ω a (ω s + ω o ))t + φ a ) (27) USB demodulation: If the rf input is a USB signal (Eq. (7)) f rf (t) = A n cos((ω c + ω n )t + φ n ), with signal frequency ω c, and we consider just a single component f n (t) = A n cos((ω c + ω n )t + φ n ), substitution for ω a in Eq. (26) gives y (+) audio (t) = A n 2 cos((ω c + ω n (ω s ω o ))t + φ n ), and if we select the tuning oscillator frequency ω s = ω c + ω o the output is y (+) audio (t) = A n 2 cos(ω nt + φ n ) which is the demodulated component, and generalizing to the sum of all such components in the full USB signal: y usb demod(t) = y (+) audio (t) = 1 2 A n cos(ω n t + φ n ). Summary: To demodulate a USB signal with frequency ω c using a low-pass filter with cut-off frequency ω o, Set the tuning oscillator to the frequency ω s = ω c + ω o, Select the audio output as the sum y (+) audio (t) = f 5(t) + f 6 (t). LSB demodulation: Similarly, if the rf input is a LSB signal (Eq. (8)) f(t) = A n cos((ω c ω n )t φ n ) 13

14 with signal frequency ω c, and we consider a single component f(t) = A n cos((ω c ω n )t φ n ), substitution for ω a in Eq. (27) gives y ( ) audio (t) = A n 2 cos((ω c ω n (ω s ω o ))t φ n ), and when the tuning oscillator frequency ω s = ω c ω o the output is y ( ) audio (t) = A n 2 cos(ω nt φ n ) which is the demodulated component. then when applied to all N such components in the full USB signal y usb demod(t) = y ( ) audio (t) = 1 2 A n cos(ω n t φ n ). Summary: To demodulate a LSB signal with frequency ω c using a low-pass filter with cut-off frequency ω o Set the tuning oscillator to the frequency ω s = ω c ω o, Select the audio output as the difference y ( ) audio (t) = f 5(t) f 6 (t). 4 DSP Implementation Issues A purely software based implementation of either the modulator or demodulator requires high-speed processing hardware, and is probably not practicable in low cost applications. 4.1 Aliasing Issues Aliasing refers to the inability of DSP systems to represent high frequency signals, and the fact that such high frequency signals will appear as if they have a lower frequency. Any DSP system operates on discrete-time signal samples at intervals of T seconds. Then the sampling frequency is defined as F sample = 1/ T Hz, or ω sample = 2πF sample rad/sec. Nyquist s sampling theorem states that for unambiguous interpretation and processing, all digital signals must represent waveforms with no component at or above the Nyquist frequency, ω N = ω sample /2. This includes external signals digitized by an A/D converter, as well as signals created within the processor, such as from digital oscillators and from the outputs from multipliers. If an attempt is made to digitize or generate a component with a frequency greater than ω N, it will be aliased to a lower frequency below ω N and will be indistinguishable from any other component at that frequency. Figure 13 shows the apparent frequency of a sinusoid sampled through an A/D converter with a sample rate of ω sample, that is ω N = ω sample /2. Below omega N the apparent frequency is accurate, but in the shaded area where ω ω N the frequency is folded down to an aliased frequency. The Weaver systems are particularly sensitive to aliasing problems, which impose severe restrictions on the bandwidth of the input and output. These restrictions differ for the modulator and demodulator. 14

15 = F F = H A J = = E C E J = I = F A I F A? J H = = E = I E C B J I E J E ),? L A H J A H I = F E C H = J A I = F A I = F A I = F A E Figure 13: Aliasing at an A/D converter. If the frequency of a sinusoidal input signal, ω in, exceeds the Nyquist rate, ω sample /2, the apparent frequency is folded down and it appears in subsequent processing as a lower frequency. Note that this is a periodic phenomenon and the above plot repeats every ω sample radians/sec Aliasing in the Weaver odulator In the modulator the maximum frequency occurs at the outputs of the second multipliers, Eqs. (16) and (17), and are translated directly to the SSB output. These frequencies must satisfy the sampling theorem, with the result is that the Weaver modulator is limited to a signal frequency ω c { ωsample /2 for LSB ω c < (28) ω sample /2 2ω o for USB where ω o is the cutoff frequency of the low pass filter. For example a system operating at the CD sampling frequency F sample = 44.1 khz is limited to generating a SSB signal with a highest frequency component below khz. This might typically be a USB signal with a 4 khz bandwidth at 18 khz, or a LSB signal with a 4kHz bandwidth at 22kHz Aliasing in the Weaver Demodulator Aliasing in the demodulator poses a slightly more complex problem because of two issues: 1) the input rf signal for the demodulator consists of the entire rf spectrum, and 2) the highest frequencies in the demodulator occur at the output of the first modulator and before the low-pass filters. (1) The analog rf input must be low-pass filtered to below the Nyquist frequency (known as pre-aliasing filtering) before being digitized at the A/D converter. This has the effect of limiting the maximum demodulation bandwidth to below ω sample /2, otherwise the whole rf spectrum will be aliased down into the demodulation bandwidth. (2) The first pair of oscillators are the high frequency tuning oscillators, ω s. Their frequency must be limited to below the Nyquist frequency ω s < ω sample/2, otherwise spurious components will be generated in the output. (3) There is an even more severe restriction if all aliasing is to be avoided (we will see that this is actually not necessary below). Eqs. (20) and (21), and Fig. 11, show the outputs 15

16 I of the first multipliers have a maximum frequency ω a + ω s, where ω a is the rf input frequency and ω s is the first oscillator frequency. But ω s ω a, which implies that aliasing in f 1 (t) and f 2 (t) will occur if 2ω s ω sample /2, or if ω s ω sample. (29) 4 For example, to avoid all aliasing, a DSP demodulator using a sampling frequency of 44.1 khz must be limited to a rf spectral width of only khz. It is not necessary, however, to eliminate all aliasing. Note that f 1 (t) and f 2 (t) are the inputs for the two low-pass filters, and therefore the filter outputs, f 3 (t) and f 4 (t), will be unaffected by aliasing provided all aliased components fall above ω o, the cut-off frequency of the filters and are therefore rejected. Figure 14 shows the situation for a rf spectrum that extends to the Nyquist frequency ω N, and an oscillator frequency ω s. The high frequency components are folded down at ω N and extend to a frequency ω N ω s. There will be no contamination of the demodulated output provided ω s ω N ω 0, (30) which is a much reduced restriction than that implied by Eq. (29). = F E J A, E C E J E 4. I E C = B J H B B E H I J K J E F E A H = F E J A B J B J = E = I F A J I I I Figure 14: Aliasing in Weaver demodulation using DSP multiplication. This figure should be compared with Fig. 11. Note that the high frequency components (above the Nyquist frequency) have been folded down to a lower frequency and will contaminate the filter outputs if ω s ω N ω o. For example, for a system sampled at the CD rate (44.1 khz) and ω o = 2kHz there will be know aliasing contamination of the output provided the oscillator frequency ω s < 20 khz. 4.2 Hybrid Implementations to Overcome Aliasing Problems Wideband DSP Based SSB Generator The limiting factor affecting the high frequency use of the Weaver (or any other) DSP modulator is the sampling and processing speed available, and the aliasing generated in the final multiplication stage. There are two approaches that might be taken: (1) Use a full DSP modulator to generate a SSB signal at as higher frequency as is practicable, then use a simple RF-mixer to up-convert that signal to the RF band. For 16

17 3 1 \$ 5 example, if processing is limited to 192 khz, generate a SSB signal at 80 khz, and then convert it to the 40m ham band ( 7 Hz). The problem with this approach that use of a simple mixer will generate an image that must be eliminated by RF band-pass filtering. (2) In Fig. 15 the second multiplier has been replaced by analog hardware that performs the same function, and is thus independent of the available DSP processing speed. The necessary sampling speed for the DSP section is therefore set by the audio bandwidth, 2ω o, and is therefore quite modest. B J F = I I B E J A H B J! B J # = E I E C = B = E J ),? I J I E J B J B E N F = I I B E J A H, )? I I J I E I J B " J B J 5 5 * K J F K J O J, 5 2, ) I L = H E = > A = = C D = = H A Figure 15: Practical implementation of a Weaver based SSB generator Wideband DSP Based SSB Receiver We saw above that the frequency span of a Weaver based receiver is severely restricted by the sampling rate (and subsequent aliasing) at the A/D converter and first multiplier stage. Almost all high-frequency SDR receivers use a hybrid approach in which the rf signal is down-converted to baseband using analog hardware that is functionally equivalent to the first multiplier stage. The structure is shown in Fig. 16. The pre-alias filters are low-pass, with a cut-off frequency set to the Nyquist frequency of the A/D converters. Note that these filters must be placed before the A/D converters, and cannot be combined with the Weaver filters. This structure allows for a much reduced DSP processing frequency and complexity. The processing has been reduced to two filters and two multiplications. The highest frequency present will be 2ω o (in the audio signals), setting a minimum sampling/processing frequency of 4ω o to prevent aliasing A Pseudo-Code Implementation of a Hybrid Weaver Demodulator Given a) A pair of identical low-pass filters with a cut-off frequency Fo (Hz) 17

18 3 1 5 H B I E C = B J H B? I I J I E I J F H A = E = I B E J A H I L = H E = > A F H A = E = I B E J A H = = C D = = H A ), ), B J B J F = I I B E J A H B E N F = I I B E J A H B! J B # J? I J I E J B J " B J \$, ) = E K J F K J O J, 5 2 Figure 16: Practical implementation of a Weaver based SSB receiver. b) A processor sampling rate Fsample (Hz) the following pseudo-code illustrates the essential steps in a Weaver demodulator as shown if Fig. 16: deltaphase = 2*pi*Fo/Fsample phase = 0 setclockinterval(1/fsample) waitclock() loop forever f1 = readad(0) // Read the two AD converters f2 = readad(1) // Avoid data skew by reading as close together as possible f3 = filtera(f1) // Low-pass filter f4 = filterb(f2) f5 = cos(phase) * f3 // ultipliers f6 = sin(phase) * f4 if (USB) then demodout = f5 + f6 // Summer else demodout = f5 - f6 phase = phase + deltaphase // Update for next sample if phase > 2*pi then phase = phase - 2*pi waitclock() // Wait for real-time clock end loop 18

19 Appendix A: Useful Trigonometric Identities Products of Sinusoidal Functions: cos(α) cos(β) = 1 2 cos(α β) + 1 cos(α + β) (A.1) 2 sin(α) sin(β) = 1 2 cos(α β) 1 cos(α + β) (A.2) 2 sin(α) cos(β) = 1 2 sin(α + β) + 1 sin(α β) (A.3) 2 cos(α) sin(β) = 1 2 sin(α + β) 1 sin(α β) (A.4) 2 These properties are useful for finding the effect of multiplying two time domain sinusoidal functions, for example using Eq (A.1): A a cos(ω a t + φ a ) cos(ω c t) = A a 2 [cos ((ω a ω c )t + φ a ) + cos ((ω a + ω c )t + φ)] which demonstrates the sum (ω a + ω c ) and difference (ω a ω c ) frequencies generated by multiplication. Sums of Angles: cos (α ± β) = cos (α) cos (β) sin (α) sin (β) (A.5) sin (α ± β) = sin (α) cos (β) ± cos (α) sin (β) (A.6) For example, the phasing method for generating a LSB signal y lsb (t) = A sin((ω s ω a )t) is: y lsb (t) = A sin((ω s ω a )t) = A sin(ω c t) cos(ω a t) A cos(ω c t) sin(ω a t) Negative Angle Formulas: cos( α) = cos(α) (A.7) sin( α) = sin(α) (A.8) These are useful for resolving the issue of apparent occurrences of negative frequencies in a sinusoid. For example if ω a > ω b in the expression A sin((ω b ω a )t + φ), the frequency is apparently negative. But by using Eq (A.8), the expression may simply be rewritten as A sin((ω b ω a )t + φ) = A sin((ω a ω b )t φ) where the frequency as written is now positive. Angle Shifts by ± π 2 (±90 ) and π (±180 ): ( sin α + π ) ( = cos (α), sin α π ) = cos (α) (A.9) ( 2 2 cos α + π ) ( = sin (α), cos α π ) = sin (α) (A.10) 2 2 sin (α ± π) = sin (α) (A.11) cos (α ± π) = cos (α) (A.12) These are useful for shifting between representations using sines and cosines. 19

20 Appendix B: Signal Processing for Spectral Components with Negative Frequencies Signal processing operations, such as multiplication, will often generate signals with an apparent negative frequency, for example f(t) = A cos( ωt+φ). The physical interpretation can be confusing, and some typical questions raised are: What the physical meaning of a negative frequency? What does a signal with a negative frequency look like on an instrument, such as an oscilloscope or a frequency analyzer? What happens if a negative frequency signal is passed through a filter whose frequency response is defined only for positive frequencies? These are, in fact, non-issues 6. It is important to remember that a quantity such as ωt+φ is simply a time-varying angle inside a trigonometric sin or cos function, and the mathematics of signal processing are completely agnostic to the sign of this angle. Any confusion may be resolved through the negative angle trigonometric formulas (Eqs. (A.7) and (A.8)) in Appendix A: cos( a) = cos(a) and sin( a) = sin(a) which allow any negative frequency component to be written in terms of an equivalent positive frequency, for example f 1 (t) = A cos( ωt + φ) A cos(ωt ψ) or f 2 (t) = A sin( ωt + ψ) A sin(ωt ψ). Such substitutions may be made at any point in the analysis, and immediately answer the second question above: a signal with a negative frequency will appear on an instrument as having a positive frequency but with a possible phase/sign change. This is often referred to as frequency folding, as shown in Fig. 17. Filtering Signals with Negative Frequency Components: The response of a linear filter (with its frequency response characteristics defined in terms of positive frequencies only) to negative frequency components is an important topic in the discussion of Weaver modulator/demodulators. Both the modulator and demodulator use low-pass filters with a passband defined from 0 ω o, yet the filter inputs necessarily contain negative frequency components from a prior multiplication. The answer is again found using the negative angle formulas. For simplicity assume ideal filters, that is the response is either unity in the passband and zero in the stopband. Write the input as a negative frequency sinusoid, and express it as the equivalent positive component: f(t) = A cos( ωt + φ) A cos(ωt φ). 6 The issues do not arise when using complex mathematics where real signal components are described by complex exponentials implicitly containing both positive and negative frequency components. 20

21 = F E J A = F E J A I F A? J H = E C Figure 17: Frequency folding of the amplitude spectra about ω = 0 to convert all components to positive frequencies only. Both the original and folded waveform representations are equally valid. The output y(t) is identical to that for a different input f (t) = A cos(ωt φ), that is { A cos(ωt φ) in the passband y(t) = 0 otherwise Expressing y(t) back to the original negative frequency input { A cos( ωt + φ) in ω is in the passband y(t) = 0 otherwise. F = I I D E C D F = I I > F = I I B J O J B J O J B J O J Figure 18: The frequency response of any linear filter is an even function of the input frequency. The result is that real filters respond symmetrically to positive and negative frequencies as indicated schematically in Fig

### CHAPTER 2! AMPLITUDE MODULATION (AM)

CHAPTER 2 AMPLITUDE MODULATION (AM) Topics 2-1 : AM Concepts 2-2 : Modulation Index and Percentage of Modulation 2-3 : Sidebands and the Frequency Domain 2-4 : Single-Sideband Modulation 2-5 : AM Power

### Amplitude Modulation, II

Amplitude Modulation, II Single sideband modulation (SSB) Vestigial sideband modulation (VSB) VSB spectrum Modulator and demodulator NTSC TV signsals Quadrature modulation Spectral efficiency Modulator

### Problems from the 3 rd edition

(2.1-1) Find the energies of the signals: a) sin t, 0 t π b) sin t, 0 t π c) 2 sin t, 0 t π d) sin (t-2π), 2π t 4π Problems from the 3 rd edition Comment on the effect on energy of sign change, time shifting

### Laboratory Assignment 5 Amplitude Modulation

Laboratory Assignment 5 Amplitude Modulation PURPOSE In this assignment, you will explore the use of digital computers for the analysis, design, synthesis, and simulation of an amplitude modulation (AM)

### Chapter 3. Amplitude Modulation Fundamentals

Chapter 3 Amplitude Modulation Fundamentals Topics Covered 3-1: AM Concepts 3-2: Modulation Index and Percentage of Modulation 3-3: Sidebands and the Frequency Domain 3-4: AM Power 3-5: Single-Sideband

### PRODUCT DEMODULATION - SYNCHRONOUS & ASYNCHRONOUS

PRODUCT DEMODULATION - SYNCHRONOUS & ASYNCHRONOUS INTRODUCTION...98 frequency translation...98 the process...98 interpretation...99 the demodulator...100 synchronous operation: ω 0 = ω 1...100 carrier

### Amplitude Modulated Systems

Amplitude Modulated Systems Communication is process of establishing connection between two points for information exchange. Channel refers to medium through which message travels e.g. wires, links, or

### EE4512 Analog and Digital Communications Chapter 6. Chapter 6 Analog Modulation and Demodulation

Chapter 6 Analog Modulation and Demodulation Chapter 6 Analog Modulation and Demodulation Amplitude Modulation Pages 306-309 309 The analytical signal for double sideband, large carrier amplitude modulation

Radio Receiver Architectures and Analysis Robert Wilson December 6, 01 Abstract This article discusses some common receiver architectures and analyzes some of the impairments that apply to each. 1 Contents

### Implementation of Digital Signal Processing: Some Background on GFSK Modulation

Implementation of Digital Signal Processing: Some Background on GFSK Modulation Sabih H. Gerez University of Twente, Department of Electrical Engineering s.h.gerez@utwente.nl Version 5 (March 9, 2016)

### Master Degree in Electronic Engineering

Master Degree in Electronic Engineering Analog and telecommunication electronic course (ATLCE-01NWM) Miniproject: Baseband signal transmission techniques Name: LI. XINRUI E-mail: s219989@studenti.polito.it

### 3.1 Introduction to Modulation

Haberlesme Sistemlerine Giris (ELE 361) 9 Eylul 2017 TOBB Ekonomi ve Teknoloji Universitesi, Guz 2017-18 Dr. A. Melda Yuksel Turgut & Tolga Girici Lecture Notes Chapter 3 Amplitude Modulation Speech, music,

### TSEK02: Radio Electronics Lecture 2: Modulation (I) Ted Johansson, EKS, ISY

TSEK02: Radio Electronics Lecture 2: Modulation (I) Ted Johansson, EKS, ISY An Overview of Modulation Techniques: chapter 3.1 3.3.1 2 Introduction (3.1) Analog Modulation Amplitude Modulation Phase and

### Communications and Signals Processing

Communications and Signals Processing Department of Communications An Najah National University 2012/2013 1 3.1 Amplitude Modulation 3.2 Virtues, Limitations, and Modifications of Amplitude Modulation

### Lecture 2: SIGNALS. 1 st semester By: Elham Sunbu

Lecture 2: SIGNALS 1 st semester 1439-2017 1 By: Elham Sunbu OUTLINE Signals and the classification of signals Sine wave Time and frequency domains Composite signals Signal bandwidth Digital signal Signal

### 6.02 Practice Problems: Modulation & Demodulation

1 of 12 6.02 Practice Problems: Modulation & Demodulation Problem 1. Here's our "standard" modulation-demodulation system diagram: at the transmitter, signal x[n] is modulated by signal mod[n] and the

### Lecture-3 Amplitude Modulation: Single Side Band (SSB) Modulation

Lecture-3 Amplitude Modulation: Single Side Band (SSB) Modulation 3.0 Introduction. 3.1 Baseband Signal SSB Modulation. 3.1.1 Frequency Domain Description. 3.1. Time Domain Description. 3. Single Tone

### Digital Signal Processing Lecture 1 - Introduction

Digital Signal Processing - Electrical Engineering and Computer Science University of Tennessee, Knoxville August 20, 2015 Overview 1 2 3 4 Basic building blocks in DSP Frequency analysis Sampling Filtering

### ECE 201: Introduction to Signal Analysis

ECE 201: Introduction to Signal Analysis Prof. Paris Last updated: October 9, 2007 Part I Spectrum Representation of Signals Lecture: Sums of Sinusoids (of different frequency) Introduction Sum of Sinusoidal

Spectrum Music 7a: Modulation Tamara Smyth, trsmyth@ucsd.edu Department of Music, University of California, San Diego (UCSD) October 3, 7 When sinusoids of different frequencies are added together, the

### Massachusetts Institute of Technology Dept. of Electrical Engineering and Computer Science Fall Semester, Introduction to EECS 2

Massachusetts Institute of Technology Dept. of Electrical Engineering and Computer Science Fall Semester, 2006 6.082 Introduction to EECS 2 Modulation and Demodulation Introduction A communication system

UNIT I FUNDAMENTALS OF ANALOG COMMUNICATION Introduction In the Microbroadcasting services, a reliable radio communication system is of vital importance. The swiftly moving operations of modern communities

### QAM Digital Communications

QAM Digital Communications By Raymond L. Barrett, Jr., PhD, PE CEO, American Research and Development, LLC www.suncam.com Copyright 00 Raymond L. Barrett, Jr. Page of 47 .0 QAM Digital Communications Introduction

### Local Oscillator Phase Noise and its effect on Receiver Performance C. John Grebenkemper

Watkins-Johnson Company Tech-notes Copyright 1981 Watkins-Johnson Company Vol. 8 No. 6 November/December 1981 Local Oscillator Phase Noise and its effect on Receiver Performance C. John Grebenkemper All

Communication Technology Laboratory Wireless Communications Group Prof. Dr. A. Wittneben ETH Zurich, ETF, Sternwartstrasse 7, 8092 Zurich Tel 41 44 632 36 11 Fax 41 44 632 12 09 Lab course Analog Part

### Sampling and Reconstruction of Analog Signals

Sampling and Reconstruction of Analog Signals Chapter Intended Learning Outcomes: (i) Ability to convert an analog signal to a discrete-time sequence via sampling (ii) Ability to construct an analog signal

### UNIT-2 Angle Modulation System

UNIT-2 Angle Modulation System Introduction There are three parameters of a carrier that may carry information: Amplitude Frequency Phase Frequency Modulation Power in an FM signal does not vary with modulation

### Multirate Digital Signal Processing

Multirate Digital Signal Processing Basic Sampling Rate Alteration Devices Up-sampler - Used to increase the sampling rate by an integer factor Down-sampler - Used to increase the sampling rate by an integer

### Real and Complex Modulation

Real and Complex Modulation TIPL 4708 Presented by Matt Guibord Prepared by Matt Guibord 1 What is modulation? Modulation is the act of changing a carrier signal s properties (amplitude, phase, frequency)

### Analog-Digital Interface

Analog-Digital Interface Tuesday 24 November 15 Summary Previous Class Dependability Today: Redundancy Error Correcting Codes Analog-Digital Interface Converters, Sensors / Actuators Sampling DSP Frequency

Modulations Analog Modulations Amplitude modulation (AM) Linear modulation Frequency modulation (FM) Phase modulation (PM) cos Angle modulation FM PM Digital Modulations ASK FSK PSK MSK MFSK QAM PAM Etc.

### Introduction. Amplitude Modulation System Angle Modulation System

Introduction Amplitude Modulation System Angle Modulation System Frequency Modulation Phase Modulation Digital Communication Elements of Information Theory Advanced Communication Techniques 1 Tools for

### Chapter 4. Part 2(a) Digital Modulation Techniques

Chapter 4 Part 2(a) Digital Modulation Techniques Overview Digital Modulation techniques Bandpass data transmission Amplitude Shift Keying (ASK) Phase Shift Keying (PSK) Frequency Shift Keying (FSK) Quadrature

### Angle Modulation. Frequency Modulation

Angle Modulation Contrast to AM Generalized sinusoid: v(t)=v max sin(ωt+φ) Instead of Varying V max, Vary (ωt+φ) Angle and Pulse Modulation - 1 Frequency Modulation Instantaneous Carrier Frequency f i

### ECEGR Lab #8: Introduction to Simulink

Page 1 ECEGR 317 - Lab #8: Introduction to Simulink Objective: By: Joe McMichael This lab is an introduction to Simulink. The student will become familiar with the Help menu, go through a short example,

### Termination Insensitive Mixers By Howard Hausman President/CEO, MITEQ, Inc. 100 Davids Drive Hauppauge, NY

Termination Insensitive Mixers By Howard Hausman President/CEO, MITEQ, Inc. 100 Davids Drive Hauppauge, NY 11788 hhausman@miteq.com Abstract Microwave mixers are non-linear devices that are used to translate

### The Digital Linear Amplifier

The Digital Linear Amplifier By Timothy P. Hulick, Ph.D. 886 Brandon Lane Schwenksville, PA 19473 e-mail: dxyiwta@aol.com Abstract. This paper is the second of two presenting a modern approach to Digital

### Elements of Communication System Channel Fig: 1: Block Diagram of Communication System Terminology in Communication System

Content:- Fundamentals of Communication Engineering : Elements of a Communication System, Need of modulation, electromagnetic spectrum and typical applications, Unit V (Communication terminologies in communication

### CHAPTER 2 DIGITAL MODULATION

2.1 INTRODUCTION CHAPTER 2 DIGITAL MODULATION Referring to Equation (2.1), if the information signal is digital and the amplitude (lv of the carrier is varied proportional to the information signal, a

### Chapter 7. Introduction. Analog Signal and Discrete Time Series. Sampling, Digital Devices, and Data Acquisition

Chapter 7 Sampling, Digital Devices, and Data Acquisition Material from Theory and Design for Mechanical Measurements; Figliola, Third Edition Introduction Integrating analog electrical transducers with

### A Low-Cost Programmable Arbitrary Function Generator for Educational Environment

Paper ID #5740 A Low-Cost Programmable Arbitrary Function Generator for Educational Environment Mr. Mani Dargahi Fadaei, Azad University Mani Dargahi Fadaei received B.S. in electrical engineering from

### EE470 Electronic Communication Theory Exam II

EE470 Electronic Communication Theory Exam II Open text, closed notes. For partial credit, you must show all formulas in symbolic form and you must work neatly!!! Date: November 6, 2013 Name: 1. [16%]

### T.J.Moir AUT University Auckland. The Ph ase Lock ed Loop.

T.J.Moir AUT University Auckland The Ph ase Lock ed Loop. 1.Introduction The Phase-Locked Loop (PLL) is one of the most commonly used integrated circuits (ICs) in use in modern communications systems.

### Lab 4: Transmission Line

1 Introduction Lab 4: Transmission Line In this experiment we will study the properties of a wave propagating in a periodic medium. Usually this takes the form of an array of masses and springs of the

### Fourier Transform And Its Application In Modulation Techniques

ourier Transform And Its Application In Modulation Techniques Mrs. Supriya Nilesh Thakur Mrs. Megha Kishor Kothawade Assistant Professor, Basic Engineering Science Department, Guru Gobind Singh College

### Agilent Vector Signal Analysis Basics. Application Note

Agilent Vector Signal Analysis Basics Application Note Table of Contents Vector signal Analysis 3 VSA measurement advantages 4 VSA measurement concepts and theory of operation 6 Data windowing leakage

### Twelve voice signals, each band-limited to 3 khz, are frequency -multiplexed using 1 khz guard bands between channels and between the main carrier

Twelve voice signals, each band-limited to 3 khz, are frequency -multiplexed using 1 khz guard bands between channels and between the main carrier and the first channel. The modulation of the main carrier

### RECOMMENDATION ITU-R F *, ** Signal-to-interference protection ratios for various classes of emission in the fixed service below about 30 MHz

Rec. ITU-R F.240-7 1 RECOMMENDATION ITU-R F.240-7 *, ** Signal-to-interference protection ratios for various classes of emission in the fixed service below about 30 MHz (Question ITU-R 143/9) (1953-1956-1959-1970-1974-1978-1986-1990-1992-2006)

### ME scope Application Note 01 The FFT, Leakage, and Windowing

INTRODUCTION ME scope Application Note 01 The FFT, Leakage, and Windowing NOTE: The steps in this Application Note can be duplicated using any Package that includes the VES-3600 Advanced Signal Processing

### Theory of Telecommunications Networks

Theory of Telecommunications Networks Anton Čižmár Ján Papaj Department of electronics and multimedia telecommunications CONTENTS Preface... 5 Introduction... 6. Mathematical models for communication channels...

### Chapter-2 SAMPLING PROCESS

Chapter-2 SAMPLING PROCESS SAMPLING: A message signal may originate from a digital or analog source. If the message signal is analog in nature, then it has to be converted into digital form before it can

### Nyquist's criterion. Spectrum of the original signal Xi(t) is defined by the Fourier transformation as follows :

Nyquist's criterion The greatest part of information sources are analog, like sound. Today's telecommunication systems are mostly digital, so the most important step toward communicating is a signal digitization.

### INTRODUCTION TO MODELLING WITH TIMS

INTRODUCTION TO MODELLING WITH TIMS model building...2 why have patching diagrams?...2 organization of experiments...3 who is running this experiment?...3 early experiments...4 modulation...4 messages...4

### Lecture 15: Introduction to Mixers

EECS 142 Lecture 15: Introduction to Mixers Prof. Ali M. Niknejad University of California, Berkeley Copyright c 2005 by Ali M. Niknejad A. M. Niknejad University of California, Berkeley EECS 142 Lecture

### Amplitude Modulation Early Radio EE 442 Spring Semester Lecture 6

Amplitude Modulation Early Radio EE 442 Spring Semester Lecture 6 f f f LO audio baseband m http://www.technologyuk.net/telecommunications/telecom_principles/amplitude_modulation.shtml AM Modulation --

### Spectrum Analysis - Elektronikpraktikum

Spectrum Analysis Introduction Why measure a spectra? In electrical engineering we are most often interested how a signal develops over time. For this time-domain measurement we use the Oscilloscope. Like

### The Interleaving Process in Digital Bandwidth Interleaving (DBI) Scopes

The Interleaving Process in Digital Bandwidth Interleaving (DBI) Scopes December, 009 Summary The signal translation aspects of Digital Bandwidth Interleaving have been explained in the White Paper Digital

### Intuitive Guide to Fourier Analysis. Charan Langton Victor Levin

Intuitive Guide to Fourier Analysis Charan Langton Victor Levin Much of this book relies on math developed by important persons in the field over the last 2 years. When known or possible, the authors have

UNIT 2 DIGITAL COMMUNICATION DIGITAL COMMUNICATION-Introduction The techniques used to modulate digital information so that it can be transmitted via microwave, satellite or down a cable pair is different

### Rec. ITU-R F RECOMMENDATION ITU-R F *,**

Rec. ITU-R F.240-6 1 RECOMMENDATION ITU-R F.240-6 *,** SIGNAL-TO-INTERFERENCE PROTECTION RATIOS FOR VARIOUS CLASSES OF EMISSION IN THE FIXED SERVICE BELOW ABOUT 30 MHz (Question 143/9) Rec. ITU-R F.240-6

### PYKC 27 Feb 2017 EA2.3 Electronics 2 Lecture PYKC 27 Feb 2017 EA2.3 Electronics 2 Lecture 11-2

In this lecture, I will introduce the mathematical model for discrete time signals as sequence of samples. You will also take a first look at a useful alternative representation of discrete signals known

### REPORT ITU-R M Adaptability of real zero single sideband technology to HF data communications

Rep. ITU-R M.2026 1 REPORT ITU-R M.2026 Adaptability of real zero single sideband technology to HF data communications (2001) 1 Introduction Automated HF communications brought a number of innovative solutions

: Receiver Architectures Outline Complex baseband equivalent of a bandpass signal Double-conversion single-quadrature (Superheterodyne) Direct-conversion (Single-conversion single-quad, homodyne, zero-)

### Fundamentals of Digital Communication

Fundamentals of Digital Communication Network Infrastructures A.A. 2017/18 Digital communication system Analog Digital Input Signal Analog/ Digital Low Pass Filter Sampler Quantizer Source Encoder Channel

### UNIT-I AMPLITUDE MODULATION (2 Marks Questions and Answers)

UNIT-I AMPLITUDE MODULATION (2 Marks Questions and Answers) 1. Define modulation? Modulation is a process by which some characteristics of high frequency carrier Signal is varied in accordance with the

### Lab 0: Introduction to TIMS AND MATLAB

TELE3013 TELECOMMUNICATION SYSTEMS 1 Lab 0: Introduction to TIMS AND MATLAB 1. INTRODUCTION The TIMS (Telecommunication Instructional Modelling System) system was first developed by Tim Hooper, then a

### 8.5 Modulation of Signals

8.5 Modulation of Signals basic idea and goals measuring atomic absorption without modulation measuring atomic absorption with modulation the tuned amplifier, diode rectifier and low pass the lock-in amplifier

### Chapter 2. Signals and Spectra

Chapter 2 Signals and Spectra Outline Properties of Signals and Noise Fourier Transform and Spectra Power Spectral Density and Autocorrelation Function Orthogonal Series Representation of Signals and Noise

### Experiment 02: Amplitude Modulation

ECE316, Experiment 02, 2017 Communications Lab, University of Toronto Experiment 02: Amplitude Modulation Bruno Korst - bkf@comm.utoronto.ca Abstract In this second laboratory experiment, you will see

### 1/14. Signal. Surasak Sanguanpong Last updated: 11 July Signal 1/14

1/14 Signal Surasak Sanguanpong nguan@ku.ac.th http://www.cpe.ku.ac.th/~nguan Last updated: 11 July 2000 Signal 1/14 Transmission structure 2/14 Transmitter/ Receiver Medium Amplifier/ Repeater Medium

### EXPERIMENT WISE VIVA QUESTIONS

EXPERIMENT WISE VIVA QUESTIONS Pulse Code Modulation: 1. Draw the block diagram of basic digital communication system. How it is different from analog communication system. 2. What are the advantages of

### CME 312-Lab Communication Systems Laboratory

Objective: By the end of this experiment, the student should be able to: 1. Demonstrate the Modulation and Demodulation of the AM. 2. Observe the relation between modulation index and AM signal envelope.

### Chapter 1 EXPERIMENT-2 DOUBLE SIDEBAND SUPPRESSED CARRIER

Chapter 1 EXPERIMENT-2 DOUBLE SIDEBAND SUPPRESSED CARRIER 1.1 Introduction 1.1.1 Objectives This experiment deals with the basic of various amplitude modulation techniques for analog communication. The

### CARRIER ACQUISITION AND THE PLL

CARRIER ACQUISITION AND THE PLL PREPARATION... 22 carrier acquisition methods... 22 bandpass filter...22 the phase locked loop (PLL)....23 squaring...24 squarer plus PLL...26 the Costas loop...26 EXPERIMENT...

HF Receivers, Part 2 Superhet building blocks: AM, SSB/CW, FM receivers Adam Farson VA7OJ View an excellent tutorial on receivers NSARC HF Operators HF Receivers 2 1 The RF Amplifier (Preamp)! Typical

### NATIONAL RADIO ASTRONOMY OBSERVATORY Charlottesville, VA

NATIONAL RADIO ASTRONOMY OBSERVATORY Charlottesville, VA ELECTRONICS DIVISION INTERNAL REPORT NO. 32 ANALYSIS OF A SINGLE-CONVERSION, ANALOG/DIGITAL SIDEBAND-SEPARATING MIXER PROTOTYPE J. R. Fisher & M.

### SAMPLING THEORY. Representing continuous signals with discrete numbers

SAMPLING THEORY Representing continuous signals with discrete numbers Roger B. Dannenberg Professor of Computer Science, Art, and Music Carnegie Mellon University ICM Week 3 Copyright 2002-2013 by Roger

### Universitas Sumatera Utara

Amplitude Shift Keying & Frequency Shift Keying Aim: To generate and demodulate an amplitude shift keyed (ASK) signal and a binary FSK signal. Intro to Generation of ASK Amplitude shift keying - ASK -

### Signal Characteristics

Data Transmission The successful transmission of data depends upon two factors:» The quality of the transmission signal» The characteristics of the transmission medium Some type of transmission medium

### EEL 4350 Principles of Communication Project 2 Due Tuesday, February 10 at the Beginning of Class

EEL 4350 Principles of Communication Project 2 Due Tuesday, February 10 at the Beginning of Class Description In this project, MATLAB and Simulink are used to construct a system experiment. The experiment

### RF/IF Terminology and Specs

RF/IF Terminology and Specs Contributors: Brad Brannon John Greichen Leo McHugh Eamon Nash Eberhard Brunner 1 Terminology LNA - Low-Noise Amplifier. A specialized amplifier to boost the very small received

### Chapter 6 CONTINUOUS-TIME, IMPULSE-MODULATED, AND DISCRETE-TIME SIGNALS. 6.6 Sampling Theorem 6.7 Aliasing 6.8 Interrelations

Chapter 6 CONTINUOUS-TIME, IMPULSE-MODULATED, AND DISCRETE-TIME SIGNALS 6.6 Sampling Theorem 6.7 Aliasing 6.8 Interrelations Copyright c 2005- Andreas Antoniou Victoria, BC, Canada Email: aantoniou@ieee.org

### EXPERIMENT 4: RC, RL and RD CIRCUITs

EXPERIMENT 4: RC, RL and RD CIRCUITs Equipment List An assortment of resistor, one each of (330, 1k,1.5k, 10k,100k,1000k) Function Generator Oscilloscope 0.F Ceramic Capacitor 100H Inductor LED and 1N4001

### Electronics Interview Questions

Electronics Interview Questions 1. What is Electronic? The study and use of electrical devices that operate by controlling the flow of electrons or other electrically charged particles. 2. What is communication?

### Massachusetts Institute of Technology Dept. of Electrical Engineering and Computer Science Spring Semester, Introduction to EECS 2

Massachusetts Institute of Technology Dept. of Electrical Engineering and Computer Science Spring Semester, 2007 6.082 Introduction to EECS 2 Lab #3: Modulation and Filtering Goal:... 2 Instructions:...

### VLSI Implementation of Digital Down Converter (DDC)

Volume-7, Issue-1, January-February 2017 International Journal of Engineering and Management Research Page Number: 218-222 VLSI Implementation of Digital Down Converter (DDC) Shaik Afrojanasima 1, K Vijaya

### I-Q transmission. Lecture 17

I-Q Transmission Lecture 7 I-Q transmission i Sending Digital Data Binary Phase Shift Keying (BPSK): sending binary data over a single frequency band Quadrature Phase Shift Keying (QPSK): sending twice

### PRINCIPLES OF COMMUNICATION SYSTEMS LAB

PRINCIPLES OF COMMUNICATION SYSTEMS LAB LAB MANUAL (EE-230-F) IV SEM ELECTRICAL AND ELECTRONICS ENGINEERING DEPARTMENT OF ELECTRICAL & ELECTRONICS DRONACHARYA COLLEGE OF ENGINEERING KHENTAWAS, GURGAON-123506

### University of Toronto Electrical & Computer Engineering ECE 316, Winter 2015 Thursday, February 12, Test #1

Name: Student No.: University of Toronto Electrical & Computer Engineering ECE 36, Winter 205 Thursday, February 2, 205 Test # Professor Dimitrios Hatzinakos Professor Deepa Kundur Duration: 50 minutes

### Design Analysis of Analog Data Reception Using GNU Radio Companion (GRC)

World Applied Sciences Journal 17 (1): 29-35, 2012 ISSN 1818-4952 IDOSI Publications, 2012 Design Analysis of Analog Data Reception Using GNU Radio Companion (GRC) Waqar Aziz, Ghulam Abbas, Ebtisam Ahmed,

### Spectrum Analysis: The FFT Display

Spectrum Analysis: The FFT Display Equipment: Capstone, voltage sensor 1 Introduction It is often useful to represent a function by a series expansion, such as a Taylor series. There are other series representations

### Operating Manual Ver 1.1

Frequency Modulation and Demodulation Trainer ST2203 Operating Manual Ver 1.1 An ISO 9001 : 2000 company 94-101, Electronic Complex Pardesipura, Indore- 452010, India Tel : 91-731- 2570301/02, 4211100

### MODELLING AN EQUATION

MODELLING AN EQUATION PREPARATION...1 an equation to model...1 the ADDER...2 conditions for a null...3 more insight into the null...4 TIMS experiment procedures...5 EXPERIMENT...6 signal-to-noise ratio...11

### Analog and Telecommunication Electronics

Politecnico di Torino Electronic Eng. Master Degree Analog and Telecommunication Electronics C5 - Synchronous demodulation» AM and FM demodulation» Coherent demodulation» Tone decoders AY 2015-16 19/03/2016-1

### An Introduction to Spectrum Analyzer. An Introduction to Spectrum Analyzer

1 An Introduction to Spectrum Analyzer 2 Chapter 1. Introduction As a result of rapidly advancement in communication technology, all the mobile technology of applications has significantly and profoundly

### The Case for Oversampling

EE47 Lecture 4 Oversampled ADCs Why oversampling? Pulse-count modulation Sigma-delta modulation 1-Bit quantization Quantization error (noise) spectrum SQNR analysis Limit cycle oscillations nd order ΣΔ

### Lecture 3 Complex Exponential Signals

Lecture 3 Complex Exponential Signals Fundamentals of Digital Signal Processing Spring, 2012 Wei-Ta Chu 2012/3/1 1 Review of Complex Numbers Using Euler s famous formula for the complex exponential The

### Fourier Transform Pairs

CHAPTER Fourier Transform Pairs For every time domain waveform there is a corresponding frequency domain waveform, and vice versa. For example, a rectangular pulse in the time domain coincides with a sinc