隨機過程與 AWGN、調變與 BER、雜訊指數與雜訊溫度、天線靈敏度與 G/T,以及非線性失真 — 附互動式視覺化Random Processes & AWGN, Modulation & BER, Noise Figure & Temperature, Antenna Sensitivity & G/T, and Nonlinear Distortion — with interactive visualizationsランダム過程と AWGN、変調と BER、雑音指数と雑音温度、アンテナ感度と G/T、そして非線形歪み — インタラクティブな可視化つき
| 雜訊型態Noise Type雑音の種類 | 來源Origin発生機構 | 主要特性Key Characteristics主な特徴 |
|---|---|---|
| 熱雜訊 (Johnson–Nyquist)Thermal (Johnson–Nyquist)熱雑音 (Johnson–Nyquist) | 束縛電荷的熱振動Thermal vibration of bound charges束縛電荷の熱振動 | 白色頻譜;存在於所有電阻、有損傳輸線與大氣中White spectrum; present in all resistors, lossy lines, atmosphere白色スペクトル。すべての抵抗、損失線路、大気中に存在する |
| 散粒雜訊Shotショット雑音 | 載子以離散方式越過接面Discrete charge carriers crossing a junction接合を横切る離散的なキャリア | $\propto$ 直流電流;在二極體與電晶體中為主要雜訊$\propto$ DC current; dominant in diodes & transistors直流電流に $\propto$。ダイオードやトランジスタで支配的 |
| 閃爍雜訊 (1/f)Flicker (1/f)フリッカ雑音 (1/f) | 半導體的表面缺陷Surface defects in semiconductors半導体の表面欠陥 | 功率 $\propto 1/f$;決定振盪器的相位雜訊底線Power $\propto 1/f$; sets oscillator phase noise floor電力は $\propto 1/f$。発振器の位相雑音フロアを決める |
| 電漿雜訊Plasmaプラズマ雑音 | 游離氣體中的隨機運動Random motion in ionized gas電離気体中のランダム運動 | 電離層傳播、接點放電火花Ionospheric propagation, sparking contacts電離層伝搬、接点のスパーク |
| 量子雜訊Quantum量子雑音 | 量子化的光子Quantized photons量子化された光子 | 在微波頻段通常可忽略 ($hf \ll kT$)Usually negligible at microwave freq ($hf \ll kT$)マイクロ波帯では通常無視できる ($hf \ll kT$) |
其中各變數為:where each variable is:ここで各変数は次のとおり:
| 符號Symbol記号 | 意義Meaning意味 | 單位Unit単位 |
|---|---|---|
| $T$ | 電阻的溫度Temperature of the resistor抵抗の温度 | 克耳文 (K)kelvin (K)ケルビン (K) |
| $B$ | 系統的頻寬Bandwidth of the systemシステムの帯域幅 | Hz |
| $f$ | 頻寬的中心頻率Center frequency of the bandwidth帯域の中心周波数 | Hz |
| $R$ | 電阻值Resistance抵抗値 | $\Omega$ |
| $h$ | 普朗克常數 $= 6.626\times10^{-34}$Planck's constant $= 6.626\times10^{-34}$プランク定数 $= 6.626\times10^{-34}$ | J·s |
| $k$ | 波茲曼常數 $= 1.380\times10^{-23}$Boltzmann's constant $= 1.380\times10^{-23}$ボルツマン定数 $= 1.380\times10^{-23}$ | J/K |
泰勒展開:$e^{hf/kT}-1 \approx hf/kT$,可得:Taylor expand: $e^{hf/kT}-1 \approx hf/kT$, yielding:テイラー展開:$e^{hf/kT}-1 \approx hf/kT$ より、次を得ます:
匹配負載得到開路電壓的一半 → 傳送的功率為:Matched load receives half the open-circuit voltage → power delivered:整合負荷は開放電圧の半分を受け取ります → 供給される電力は:
含雜訊的電阻 $R$ 可模型化為A noisy resistor $R$ is modeled as a雑音を持つ抵抗 $R$ は 戴維寧等效電路Thévenin equivalentテブナン等価回路:一個承載全部雜訊的理想電壓源 $v_n^s$,與無雜訊的內電阻 $R_s = R$ 串聯。這與任何其他戴維寧電源完全相同——內電阻就是往電源方向看進去的阻抗。: an ideal voltage source $v_n^s$ (carrying all the noise) in series with a noise-free internal resistance $R_s = R$. This is identical to any other Thévenin source — the internal resistance is simply the impedance looking back into the source.としてモデル化されます。すなわち、すべての雑音を担う理想電圧源 $v_n^s$ と、雑音のない内部抵抗 $R_s = R$ の直列接続です。これは他のテブナン電源とまったく同じで、内部抵抗とは電源側を見込んだインピーダンスにほかなりません。
當接上負載 $R_L$ 時,會形成When the load $R_L$ is connected, a負荷 $R_L$ を接続すると、 分壓器voltage divider分圧回路 。負載上的電壓為:forms. The voltage across the load is:が形成されます。負荷にかかる電圧は次のとおりです:
一般情況下傳送到負載的功率為:The power delivered to the load in general:一般に負荷へ供給される電力は次のとおりです:
也可以用比值 $r = R_L/R_s$ 來表示:This can also be written in terms of the ratio $r = R_L/R_s$:比 $r = R_L/R_s$ を用いて次のように書くこともできます:
根據By theいわゆる 最大功率傳輸定理maximum power transfer theorem最大電力伝達の定理,$P_{load}$ 在 $R_L = R_s$ 時(即 $r = 1$)達到最大。取導數 $\frac{d}{dr}\!\left[\frac{r}{(1+r)^2}\right] = \frac{1-r}{(1+r)^3} = 0$ 可確認峰值位於 $r=1$。, $P_{load}$ is maximized when $R_L = R_s$ (i.e., $r = 1$). Taking the derivative $\frac{d}{dr}\!\left[\frac{r}{(1+r)^2}\right] = \frac{1-r}{(1+r)^3} = 0$ confirms the peak at $r=1$.により、$P_{load}$ は $R_L = R_s$ のとき(すなわち $r = 1$)最大になります。微分 $\frac{d}{dr}\!\left[\frac{r}{(1+r)^2}\right] = \frac{1-r}{(1+r)^3} = 0$ を計算すれば、ピークが $r=1$ にあることが確認できます。
在一般功率表示式中令 $R_L = R_s = R$(匹配):Set $R_L = R_s = R$ (matched) in the general power expression:一般的な電力の式で $R_L = R_s = R$(整合)とおきます:
這個Theこの 4 來自 $(R_s + R_L)^2 = (R + R)^2 = (2R)^2 = 4R^2$。物理上:分壓器只讓負載得到開路電壓的comes from $(R_s + R_L)^2 = (R + R)^2 = (2R)^2 = 4R^2$. Physically: the voltage divider gives the load onlyは $(R_s + R_L)^2 = (R + R)^2 = (2R)^2 = 4R^2$ から来ます。物理的には、分圧回路は負荷に開放電圧の 一半half半分 ($v_n = v_n^s/2$),將這一半平方後便得到開路電壓平方的 $1/4$。the open-circuit voltage ($v_n = v_n^s/2$), and squaring that half gives $1/4$ of the open-circuit voltage squared.($v_n = v_n^s/2$)しか与えず、その半分を二乗すると開放電圧の二乗の $1/4$ になります。
現在代入 $v_n^s = \sqrt{4kTBR}$(來自瑞立–金斯的結果):Now substitute $v_n^s = \sqrt{4kTBR}$ (from the Rayleigh–Jeans result):ここで $v_n^s = \sqrt{4kTBR}$(レイリー–ジーンズの結果)を代入します:
分子的因數 4(來自普朗克/瑞立–金斯)與分母的因數 4(來自匹配分壓)恰好相消——這就是為什麼最大可用雜訊功率 $kTB$ 與電阻 $R$ 無關。The factor of 4 in the numerator (from Planck/Rayleigh–Jeans) and the factor of 4 in the denominator (from the matched voltage divider) cancel exactly — which is why the maximum available noise power $kTB$ is independent of resistance $R$.分子の係数 4(プランク/レイリー–ジーンズ由来)と分母の係数 4(整合した分圧回路由来)はちょうど打ち消し合います。これが、最大利用可能雑音電力 $kTB$ が抵抗 $R$ に依存しない理由です。
下圖顯示這個關係。將滑鼠移到曲線上可讀出精確數值。紅點標示匹配條件 $R_L = R_s$,此時傳送的功率最大($= kTB$):The chart below shows this relationship. Hover over the curve to read exact values. The red dot marks the matched condition $R_L = R_s$ where maximum power ($= kTB$) is delivered:下のグラフはこの関係を示します。曲線上にカーソルを合わせると正確な値が読み取れます。赤い点は整合条件 $R_L = R_s$ を示し、このとき供給電力が最大($= kTB$)になります:
最大功率傳輸定理正是本章始終假設The maximum power transfer theorem is why最大電力伝達の定理があるからこそ、本章では常に 阻抗匹配impedance matchingインピーダンス整合 ($Z_L = Z_0 = 50\;\Omega$)的原因。任何失配都會使傳送的雜訊功率變小——但訊號功率同樣變小。當我們寫下 $P_n = kTB$ 時,其實已隱含假設完美匹配。相同的原理也適用於放大器輸入端、天線饋線,以及接收鏈中的每一個接點。($Z_L = Z_0 = 50\;\Omega$) is always assumed throughout this chapter. Any mismatch means less noise power is delivered — but also less signal power. When we write $P_n = kTB$, we are implicitly assuming perfect matching. The same principle applies to amplifier inputs, antenna feedlines, and every junction in the receiver chain.($Z_L = Z_0 = 50\;\Omega$)を仮定します。不整合があれば供給される雑音電力は小さくなりますが、信号電力も同じだけ小さくなります。$P_n = kTB$ と書くとき、そこには完全整合という前提が暗黙に含まれています。同じ原理は増幅器の入力、アンテナ給電線、受信チェーンのあらゆる接続点にも当てはまります。
題目:Problem:問題: 一個 75 Ω 電阻在 $T = 350$ K 下,透過頻寬 $B = 2$ MHz 的理想帶通濾波器接到匹配的 75 Ω 負載。求 (a) 開路雜訊電壓的有效值,(b) 傳送到負載的雜訊功率,(c) 雙邊功率頻譜密度。A 75 Ω resistor at $T = 350$ K is connected to a matched 75 Ω load through an ideal bandpass filter with bandwidth $B = 2$ MHz. Find (a) the rms open-circuit noise voltage, (b) the noise power delivered to the load, and (c) the two-sided power spectral density.75 Ω の抵抗を $T = 350$ K に置き、帯域幅 $B = 2$ MHz の理想帯域通過フィルタを介して整合した 75 Ω 負荷に接続します。(a) 開放雑音電圧の実効値、(b) 負荷へ供給される雑音電力、(c) 両側電力スペクトル密度を求めなさい。
(a) $v_n^s = \sqrt{4kTBR} = \sqrt{4 \times 1.38\times10^{-23} \times 350 \times 2\times10^6 \times 75}$
$= \sqrt{4 \times 1.38\times10^{-23} \times 350 \times 2\times10^6 \times 75} = \sqrt{2.898\times10^{-12}} = 1.70\;\mu\text{V (rms)}$
(b) $P_n = kTB = 1.38\times10^{-23} \times 350 \times 2\times10^6 = 9.66\times10^{-15}$ W $= -110.1$ dBm
(注意:與 75 Ω 的電阻值無關!)(Note: independent of the 75 Ω resistance!)(注意:75 Ω という抵抗値には依存しません!)
(c) $S_n = kT/2 = 1.38\times10^{-23}\times350/2 = 2.415\times10^{-21}$ W/Hz(雙邊)$S_n = kT/2 = 1.38\times10^{-23}\times350/2 = 2.415\times10^{-21}$ W/Hz (two-sided)$S_n = kT/2 = 1.38\times10^{-23}\times350/2 = 2.415\times10^{-21}$ W/Hz(両側)
PDF 在這裡指的是here stands forはここでは 機率密度函數Probability Density Function確率密度関数 ——與 PDF 檔案毫無關係!它描述的是— it has nothing to do with PDF files! It describesを意味し、PDF ファイルとは何の関係もありません。これは、 有多大的機率how likelyどれくらいの確率で 讓雜訊電壓在任一瞬間取到某個特定的值。the noise voltage is to take any particular value at any instant.雑音電圧が任意の瞬間に特定の値をとるかを表します。
雜訊電壓 $v_n$ 是隨機的——任一瞬間可能為正或負、可大可小。PDF $f(v_n)$ 告訴你在任一範圍內找到 $v_n$ 的機率:具體而言,$f(v_n)\,dv_n$ 表示雜訊電壓落在 $v_n$ 與 $v_n + dv_n$ 之間的機率。The noise voltage $v_n$ is random — at any moment it could be positive or negative, large or small. The PDF $f(v_n)$ tells you the probability of finding $v_n$ in any range: specifically, $f(v_n)\,dv_n$ gives the probability that the noise voltage falls between $v_n$ and $v_n + dv_n$.雑音電圧 $v_n$ はランダムで、どの瞬間にも正にも負にも、大きくも小さくもなり得ます。PDF $f(v_n)$ は任意の範囲に $v_n$ が入る確率を与えます。具体的には $f(v_n)\,dv_n$ が、雑音電圧が $v_n$ と $v_n + dv_n$ の間に入る確率を表します。
實體電阻可視為許多微小電阻的串聯,每一個都貢獻一個微小且彼此獨立的隨機電壓。根據A physical resistor can be thought of as a series connection of many tiny resistors, each contributing a small independent random voltage. By the実際の抵抗は、多数の微小な抵抗が直列に接続されたものと考えられ、それぞれが小さく互いに独立なランダム電圧を寄与します。 中央極限定理Central Limit Theorem中心極限定理,大量獨立隨機變數之和必定收斂到高斯(鐘形)分布——與各別來源本身的分布無關。因此熱雜訊具有高斯 PDF:, the sum of many independent random variables always converges to a Gaussian (bell-curve) distribution — regardless of the distribution of each individual source. Therefore, thermal noise has the Gaussian PDF:により、多数の独立な確率変数の和は、個々の分布によらず必ずガウス(釣鐘型)分布に収束します。したがって熱雑音はガウス型の PDF をもちます:
其withであり、 均值為零zero mean平均は零 (鐘形曲線以 $v_n = 0$ 為中心)。(the bell curve is centered at $v_n = 0$).です(釣鐘型曲線は $v_n = 0$ を中心とします)。
均值為 $\mu$、標準差為 $\sigma$ 的隨機變數 $X$,其完整常態分布寫成 $X \sim \mathcal{N}(\mu,\,\sigma^2)$,PDF 為:The full normal distribution for a random variable $X$ with mean $\mu$ and standard deviation $\sigma$ is written $X \sim \mathcal{N}(\mu,\,\sigma^2)$ with PDF:平均 $\mu$、標準偏差 $\sigma$ をもつ確率変数 $X$ の完全な正規分布は $X \sim \mathcal{N}(\mu,\,\sigma^2)$ と書かれ、その PDF は次のとおりです:
$$f(x) = \frac{1}{\sigma\sqrt{2\pi}}\,e^{-\dfrac{(x-\mu)^2}{2\sigma^2}}$$對熱雜訊而言 $\mu = 0$(無直流偏移),因此公式簡化為上式。For thermal noise, $\mu = 0$ (no DC offset), so the formula simplifies to the equation above. The熱雑音では $\mu = 0$(直流オフセットなし)なので、式は上のものに簡略化されます。 標準常態分布standard normal標準正規分布 是 $\mu=0,\,\sigma=1$ 的特例:is the special case $\mu=0,\,\sigma=1$:は $\mu=0,\,\sigma=1$ の特別な場合です:
$$f(x) = \frac{1}{\sqrt{2\pi}}\,e^{-x^2/2} \qquad Z \sim \mathcal{N}(0,\,1)$$| 性質Property性質 | 數值Value値 |
|---|---|
| 均值Mean平均 | $\mu$(鐘形的中心)$\mu$ (center of bell)$\mu$(釣鐘の中心) |
| 標準差Standard deviation標準偏差 | $\sigma$(鐘形的寬度)$\sigma$ (width of bell)$\sigma$(釣鐘の幅) |
| 變異數Variance分散 | $\sigma^2$ |
| 總面積Total area全面積 | $\displaystyle\int_{-\infty}^{\infty}f(x)\,dx = 1$ |
| Symmetry | $f(\mu+x) = f(\mu-x)$ — bell is symmetric about $\mu$ |
| Peak value | $\dfrac{1}{\sigma\sqrt{2\pi}}$ at $x = \mu$ |
Connection to BER: BPSK noise $n \sim \mathcal{N}(0,\sigma^2)$, so the error probability is the tail integral beyond the threshold $v_0$:
$$P_e = \int_{v_0}^{\infty} \frac{1}{\sigma\sqrt{2\pi}}\,e^{-n^2/2\sigma^2}\,dn = \frac{1}{2}\,\mathrm{erfc}\!\left(\frac{v_0}{\sigma\sqrt{2}}\right)$$The erfc is simply the Gaussian tail integral — which is why it appears in every BER formula.
| Parameter | Symbol | Physical Meaning |
|---|---|---|
| Variance | $\sigma^2$ | Noise power for band-limited noise: $\sigma^2 = n_0 B$ |
| Standard deviation | $\sigma$ | RMS noise voltage (when $R = 1\;\Omega$); wider $\sigma$ = noisier |
| Mean | $0$ | No DC offset — noise averages to zero over time |
For a Gaussian distribution: 68% of the time $v_n$ falls within $\pm\sigma$, 95% within $\pm 2\sigma$, 99.7% within $\pm 3\sigma$. Noise exceeds $3\sigma$ only 0.3% of the time — but in a communication system running at millions of bits per second, even rare events cause bit errors.
The two-sided PSD of thermal noise is:
This PSD is flat — it has the same value at every frequency (within the Rayleigh–Jeans approximation). A signal with flat PSD is called white noise, by analogy with white light which contains all colors equally.
The total noise power is recovered by integrating the PSD over bandwidth:
The autocorrelation function $R(\tau)$ measures how similar a noise signal is to a time-shifted copy of itself:
If the noise at time $t$ is completely unrelated to the noise at time $t+\tau$, then $R(\tau) = 0$. White noise has exactly this property — samples at any two different times are uncorrelated.
The Wiener–Khinchin theorem states that the PSD and autocorrelation are a Fourier transform pair:
We start with ideal white noise: $S(\omega) = n_0/2$ (constant for all $\omega$).
Step 1 — Substitute into the Wiener–Khinchin integral:
Step 2 — Pull out the constant ($n_0/2$ does not depend on $\omega$):
Step 3 — Recognize the standard Fourier identity:
Using Euler's formula, expand the integrand $e^{j\omega\tau} = \cos(\omega\tau) + j\sin(\omega\tau)$:
The $\sin$ part: always zero by symmetry
$\sin(\omega\tau)$ is an odd function of $\omega$ (since $\sin(-x) = -\sin(x)$). When you integrate any odd function over the symmetric interval $(-\infty, +\infty)$, the negative-$\omega$ half exactly cancels the positive-$\omega$ half:
The $\cos$ part when $\tau \neq 0$: perfect cancellation
$\cos(\omega\tau)$ oscillates between $+1$ and $-1$ as $\omega$ varies. Think of it as a wave along the $\omega$-axis:
Every positive half-cycle is immediately followed by a negative half-cycle of exactly equal area. Over $(-\infty,+\infty)$, they all pair up and cancel:
The $\cos$ part when $\tau = 0$: no oscillation
$\cos(\omega\cdot 0) = \cos(0) = 1$ for every $\omega$. The integrand is just the constant $1$ — there is no oscillation at all, so nothing cancels:
Summary:
| Integrand | Oscillates? | Cancellation? | Result | |
|---|---|---|---|---|
| $\tau \neq 0$ | $\cos(\omega\tau)$ | Yes — every $+$ half-cycle has a matching $-$ half-cycle | Perfect cancellation | $0$ |
| $\tau = 0$ | $\cos(0) = 1$ | No — flat constant | Nothing to cancel | $\infty$ |
Zero everywhere except infinite at one point — that is exactly the Dirac delta function $\delta(\tau)$. At $\tau = 0$ everything adds constructively (infinite result); at any $\tau \neq 0$ the oscillations destroy themselves completely (zero result).
The Dirac delta function is defined by two properties: $\delta(\tau) = 0$ for all $\tau \neq 0$, and $\int_{-\infty}^{\infty}\delta(\tau)\,d\tau = 1$.
From these, the sifting property follows directly. If $\delta(\tau)$ is zero everywhere except at $\tau = 0$, then multiplying it by any function $g(\tau)$ only "survives" at $\tau = 0$:
$$\int_{-\infty}^{\infty} \delta(\tau)\cdot g(\tau)\,d\tau$$Since $\delta(\tau) = 0$ for $\tau \neq 0$, the only point contributing to this integral is $\tau = 0$. At that point, $g(\tau)$ has the value $g(0)$, so we can pull it out as a constant:
The delta function "sifts out" or "samples" the value of $g$ at the origin — hence the name.
Apply the sifting property with $g(\tau) = e^{-j\omega\tau}$:
Step by step: (1) $\delta(\tau) = 0$ for $\tau \neq 0$, so only $\tau = 0$ contributes. (2) Evaluate $e^{-j\omega\tau}$ at $\tau = 0$: $e^{-j\omega\cdot 0} = e^0 = 1$. (3) Result is $1$, regardless of $\omega$.
This confirms $\delta(\tau) \leftrightarrow 1$ is a Fourier pair: a delta in time ↔ a constant in frequency.
Step 4 — Combine:
The entire derivation reduces to recognizing the delta function identity — everything else is just pulling out a constant. Summarizing the Fourier pair:
$S(\omega) = \dfrac{n_0}{2}$
Constant (flat PSD)
$R(\tau) = \dfrac{n_0}{2}\,\delta(\tau)$
Delta (uncorrelated)
This is a Dirac delta function — infinite at $\tau = 0$ and zero everywhere else:
| Time shift | $R(\tau)$ | Physical meaning |
|---|---|---|
| $\tau = 0$ | $R(0) = \frac{n_0}{2}\delta(0) \to \infty$ | Total noise power (signal compared with itself) |
| $\tau \neq 0$ | $R(\tau) = 0$ | Perfectly uncorrelated at any time shift |
Setting $\tau = 0$ gives infinite total power — which is physically impossible! The issue is that $S(\omega) = n_0/2$ being constant at all frequencies (out to infinity) is only a mathematical idealization. In reality, the Rayleigh–Jeans approximation breaks down at very high frequencies, and the PSD must roll off. The flat-PSD model is valid only over a finite, though very wide, frequency band.
For noise limited to a finite bandwidth $B$, the autocorrelation becomes a sinc function (not a delta), and the total power at $\tau = 0$ is finite:
The two-sided PSD $S_n(f) = n_0/2$ is defined over both positive and negative frequencies. For a system with one-sided bandwidth $B$ (the physical bandwidth you'd see on a spectrum analyzer), the two-sided frequency range extends from $-B$ to $+B$:
The $2$ in the numerator of $2B$ cancels the $2$ in the denominator of $n_0/2$, giving $n_0 B = kTB$.
Alternatively, using the one-sided PSD $S_n(f) = n_0$ (defined only for $f \geq 0$), you integrate over just $B$ and get the same result directly:
$\displaystyle R(0) = \int_0^{B} n_0\,df = n_0\cdot B = kTB$
The factor of 2 is purely a bookkeeping difference between the two PSD conventions — the physical noise power is always $kTB$.
We use the "white" (flat PSD) model because it's an excellent approximation within the system bandwidth. But we must always integrate over the finite bandwidth $B$ to get physically meaningful power. The delta function $\delta(\tau)$ and infinite power are mathematical artifacts of extending the flat-PSD model to infinite bandwidth — they never occur in real systems.
Combining "white" (flat PSD) + "Gaussian" (amplitude PDF) gives AWGN — Additive White Gaussian Noise:
Noise adds to the signal: $r(t) = s(t) + v_n(t)$
Flat PSD → all frequencies contribute equally
Amplitude follows the bell-curve PDF above
AWGN is the standard noise model used in all BER calculations in this chapter
By definition, AWGN has a flat two-sided PSD at every frequency:
$$S(f) = \frac{n_0}{2} \quad \text{for all } f \in (-\infty,\, +\infty)$$The $\frac{1}{2}$ is always present because this is the two-sided convention — the total noise density $n_0$ is split equally between positive and negative frequencies. The value of $n_0$ depends on the physical source:
$$n_0 = kT \quad \Rightarrow \quad \frac{n_0}{2} = \frac{kT}{2} \approx 2\times10^{-21}\;\text{W/Hz at }T_0=290\;\text{K}$$| Property | Expression | Note |
|---|---|---|
| Two-sided PSD | $S(f) = n_0/2$ | Defined for all $f$, positive and negative |
| One-sided PSD | $S(f) = n_0$ | Defined for $f \geq 0$ only — double the two-sided value |
| Noise power in BW $B$ | $N = \frac{n_0}{2}\times 2B = n_0 B = kTB$ | Two-sided integral over $[-B, +B]$ |
| Noise power in BW $B$ | $N = n_0 \times B = kTB$ | One-sided integral over $[0, B]$ — same result |
The two-sided PSD $n_0/2$ integrates over a bandwidth of $2B$ (from $-B$ to $+B$): $\frac{n_0}{2}\times 2B = n_0 B$. The one-sided PSD $n_0$ integrates over just $B$ (from $0$ to $B$): $n_0 \times B = n_0 B$. The factor of 2 in the PSD and the factor of 2 in the bandwidth always cancel — physical noise power is always $n_0 B = kTB$ regardless of which convention you use.
Problem: A receiver has a system noise temperature of 600 K and a bandwidth of 5 MHz. Find the total noise power and the one-sided PSD $n_0$.
$n_0 = kT = 1.38\times10^{-23}\times 600 = 8.28\times10^{-21}$ W/Hz
$N = n_0 B = 8.28\times10^{-21}\times 5\times10^6 = 4.14\times10^{-14}$ W $= -103.8$ dBm
Power is defined as a time-domain quantity — $P = \lim_{T\to\infty}\frac{1}{T}\int v^2(t)\,dt$. Yet we routinely compute it by integrating the PSD over frequency: $P = \int S(f)\,df$. Why are these two expressions equal? The answer is Parseval's theorem.
For a noise signal $v(t)$, average power (across a $1\;\Omega$ load) can be computed three equivalent ways:
| Domain | Expression | What It Uses |
|---|---|---|
| Time | $\displaystyle P = \lim_{T\to\infty}\frac{1}{T}\int_0^T v^2(t)\,dt$ | Waveform directly |
| Frequency (PSD) | $\displaystyle P = \int_{-\infty}^{\infty} S(f)\,df$ | Power per Hz — measurable with spectrum analyzer |
| Statistical | $\displaystyle P = E[v^2] = \sigma^2$ (for zero-mean noise) | Variance of the amplitude distribution |
For an ergodic stationary random process — which thermal noise is — all three give the same numerical result.
Variance is defined as the average squared deviation from the mean:
$$\text{Var}(v_n) = E[(v_n - \mu)^2]$$Expanding the square and using linearity of expectation:
$$= E[v_n^2] - 2\mu\,E[v_n] + \mu^2 = E[v_n^2] - \mu^2$$For noise, $\mu = E[v_n] = 0$ (zero mean — noise averages to zero over time), so:
$$\sigma^2 = \text{Var}(v_n) = E[v_n^2] - 0 = E[v_n^2]$$And $E[v_n^2]$ is the mean square value, which equals average power across $1\,\Omega$ by the time-domain definition. So all three are the same:
$$\underbrace{\sigma^2}_{\text{variance}} = \underbrace{E[v_n^2]}_{\text{mean square}} = \underbrace{P}_{\text{noise power}}$$For a deterministic signal $v(t)$ with Fourier transform $V(f)$, total energy is preserved across the transform:
The left side is energy computed in time. The right side is the same energy computed from the spectrum. $|V(f)|^2$ is the Energy Spectral Density (J/Hz).
Write $|v(t)|^2 = v(t)\cdot v^*(t)$, then substitute the inverse Fourier transform for one factor of $v(t)$:
The swap of integration order (Fubini's theorem) is justified for square-integrable signals. The inner bracket in the last step is exactly $V^*(f)$ by the definition of the Fourier transform.
Thermal noise has infinite total energy, so Parseval in its energy form diverges. Instead, we truncate the signal to a window of duration $T$ and define:
where $V_T(f)$ is the Fourier transform of the windowed signal $v_T(t)$. Dividing Parseval's energy identity by $T$ and taking the limit:
A prism splits white light into a spectrum of colors. Parseval's theorem says: the total brightness of the white beam equals the sum of brightnesses of every color band. Whether you measure the combined beam directly (time domain) or sum all the spectral slices (frequency domain), you get the same total power. The Fourier transform is the mathematical prism.
The PSD is also the Fourier transform of the autocorrelation function $R_v(\tau) = E[v(t)v(t+\tau)]$:
Setting $\tau = 0$: $R_v(0) = E[v^2] = P_{avg}$. And by the inverse Fourier transform at $\tau=0$: $R_v(0) = \int S(f)\,df$. These are consistent — confirming again that $\int S(f)\,df = P_{avg}$, this time via the autocorrelation route.
Integrating PSD over frequency is computing time-averaged power. It is not a new definition — it is a direct consequence of Parseval's theorem applied to power signals. The time-domain average and the frequency-domain PSD integral are two views of the same quantity, linked by the Fourier transform.
Problem: Thermal noise from a $50\;\Omega$ resistor at $T=290$ K passes through an ideal bandpass filter of bandwidth $B = 1$ MHz centered at $f_0 = 2.4$ GHz. (a) Write the two-sided PSD $S(f)$. (b) Use Parseval's theorem to find the total average noise power. (c) Verify using $P_n = kTB$.
(a) Within the filter passband $[f_0 - B/2,\; f_0 + B/2]$ (and its negative-frequency image), $S(f) = kT/2 = n_0/2$. Outside the passband $S(f) = 0$.
(b) By Parseval / definition of PSD:
$P = \displaystyle\int_{-\infty}^{\infty} S(f)\,df = \frac{n_0}{2}\times 2B = n_0 B = kTB$
(Factor of 2 because the two-sided PSD integrates over both positive and negative frequency bands of width $B$.)
(c) Direct check: $P_n = kTB = 1.38\times10^{-23}\times290\times10^6 = 4.0\times10^{-15}$ W $= -113.9$ dBm. ✓
The Parseval integral and the $kTB$ formula give identical results — confirming the equivalence.
SC = Suppressed Carrier. To understand why, compare standard AM with DSB-SC:
In conventional AM (double-sideband with carrier), the transmitted signal is:
The $A_c\cos(2\pi f_c t)$ term is the carrier — it carries zero information yet consumes a large fraction of the total transmit power. DSB-SC suppresses this term entirely, transmitting only the sidebands:
| DSB (with carrier) | DSB-SC | |
|---|---|---|
| Carrier transmitted | Yes | No — suppressed |
| Power efficiency | Low — carrier wastes power | High — all power in sidebands |
| Demodulation | Simple envelope detector | Coherent local oscillator required |
| Bandwidth | $2f_m$ | $2f_m$ (same) |
Removing the carrier saves significant transmit power, but the receiver loses its phase reference. A coherent local oscillator — phase-locked to the original carrier frequency — is now required at the receiver to demodulate. SSB takes this one step further: it also removes one of the two sidebands, halving the bandwidth as well, but requiring an even more precise receiver.
The complete SSB communication system has three stages: modulator, channel, and demodulator. Understanding what each block does to the signal and noise is essential for the SNR analysis.
| Block | Function | Signal At Output |
|---|---|---|
| Mixer × (modulator) | Multiplies $m(t)$ by carrier $\cos(2\pi f_{IF}t)$ | DSB signal: both upper and lower sidebands around $f_{IF}$ |
| Sideband filter | Removes one sideband | SSB signal: half the bandwidth of DSB |
| + $n_0/2$ (channel) | AWGN added by the channel | SSB signal + white noise |
| IF filter | Bandpass filter: selects desired signal, limits noise BW | $v_i(t) = A\cos(\omega_{IF}+\omega_m)t + v_n(t)$ with power $S_i, N_i$ |
| Mixer × (demod) | Multiplies $v_i(t)$ by $\cos(2\pi f_{IF}t)$ → translates to baseband | Baseband signal + noise + double-frequency terms |
| Baseband filter | Low-pass filter removes $2f_{IF}$ terms | $v_o(t) = \frac{A}{2}\cos\omega_m t + \frac{1}{2}x(t)$ |
The noise at the IF filter output is represented using its in-phase and quadrature components:
where $x(t)$ and $y(t)$ are band-limited, independent, zero-mean random processes with equal power: $E\{x^2\} = E\{y^2\} = N_i$.
The received signal plus noise enters the demodulator:
After mixing with $\cos\omega_{IF}t$ and low-pass filtering (removing all $2\omega_{IF}$ terms):
Both the signal and noise are scaled by the same factor of $1/2$:
| Quantity | Expression | How it changes |
|---|---|---|
| Input signal power $S_i$ | $A^2/2$ | |
| Output signal power $S_o$ | $(A/2)^2/2 = A^2/8$ | $= S_i/4$ |
| Input noise power $N_i$ | $E\{x^2\} = E\{y^2\}$ | |
| Output noise power $N_o$ | $E\{[x(t)/2]^2\} = N_i/4$ | $= N_i/4$ |
The DSB-SC system is similar to SSB but without the sideband filter — both sidebands are transmitted. The IF filter bandwidth is $2f_m$ (twice that of SSB). Note there is no sideband filter in the modulator:
Key differences from SSB: no sideband filter in the modulator (both sidebands transmitted), and the IF filter has bandwidth $2f_m$ instead of $f_m$. The input signal has two components at $\omega_{IF}\pm\omega_m$, each with amplitude $A/\sqrt{2}$. After mixing and filtering:
| Quantity | Expression | How it changes |
|---|---|---|
| Input signal power $S_i$ | $\frac{1}{2}(A/\sqrt{2})^2 + \frac{1}{2}(A/\sqrt{2})^2 = A^2/2$ | |
| Output signal power $S_o$ | $\frac{1}{2}(A/\sqrt{2})^2 = A^2/4$ | $= S_i/2$ |
| Output noise power $N_o$ | $N_i/4$ | $= N_i/4$ (same as SSB) |
In DSB-SC, both sidebands carry the same signal information. At the demodulator, these two copies add coherently (voltages add → power doubles). But the noise from each sideband is independent, so noise adds only incoherently (powers add, not voltages). The result: signal power is doubled relative to noise, giving a 3 dB advantage. The trade-off is that DSB-SC requires twice the bandwidth of SSB.
To truly understand why SSB preserves SNR while DSB-SC improves it by 3 dB, you need to see what happens to noise at the mixer — the mirror image folding effect.
Before the mixer: the signal sits at IF frequency $f_{IF}$. Noise with two-sided PSD $n_0/2$ surrounds it symmetrically — both the upper sideband ($f_{IF}+f_m$) and lower sideband ($f_{IF}-f_m$) contain independent noise:
After the mixer: multiplying by $\cos(2\pi f_{IF}t)$ shifts the entire spectrum both down and up by $f_{IF}$. The upper sideband translates down to baseband at $+f_m$. The lower sideband translates to $-f_m$, which mirrors (folds) onto $+f_m$. Both land on the same baseband frequencies:
Since the two sideband noise sources are independent (uncorrelated), their powers add:
This folding is why the two-sided PSD ($n_0/2$) matters — after mixing, both sides fold together to give $n_0$ per unit bandwidth at baseband.
The noise folds identically in both cases. The difference is what happens to the signal:
| Signal at upper sideband | Signal at lower sideband | After mixing (signal) | After mixing (noise) | SNR change | |
|---|---|---|---|---|---|
| SSB | Signal present | Empty (filtered out) | Signal $\times 1$ only | Noise from both sides folds: $\times 2$ | No improvement |
| DSB-SC | Signal copy 1 | Signal copy 2 (same info) | Two copies add coherently: voltage $\times 2$ → power $\times 4$ | Noise from both sides folds: $\times 2$ | $\times 2$ → 3 dB better |
The mixer always folds noise from both sidebands into baseband — this is unavoidable. In SSB, only one sideband has signal, so the signal doesn't benefit from the folding. In DSB-SC, the signal is in both sidebands and adds coherently (voltages, not powers), beating the noise folding by a factor of 2.
This is also why the two-sided PSD ($n_0/2$) is essential for demodulation analysis — you must account for noise from both sidebands folding onto the same baseband frequencies.
Problem: A communication link has an input SNR of 25 dB. Compare the output SNR for (a) SSB and (b) DSB-SC demodulation.
(a) SSB: $\text{SNR}_{out} = \text{SNR}_{in} = 25$ dB (no change)
(b) DSB-SC: $\text{SNR}_{out} = \text{SNR}_{in} + 3 = 28$ dB
The DSB-SC gives 3 dB better output SNR, at the cost of requiring twice the bandwidth.
DSB adds the carrier term $A_c\cos(\omega_{IF}t)$ back into the transmitted signal. The modulator no longer suppresses it — the carrier goes through alongside both sidebands:
The key difference: the demodulator is a simple envelope detector (diode + RC) — no coherent local oscillator needed. The carrier provides a built-in phase reference, so the envelope of $v_i(t)$ directly tracks $m(t)$. The DC block afterwards removes the DC offset $A_c/2$ left by the carrier.
The envelope detector has three components — diode, capacitor, and resistor — each playing a distinct role:
| Component | Role | What happens |
|---|---|---|
| Diode D | Half-wave rectifier | Blocks negative half-cycles — only positive peaks pass through |
| Capacitor C | Peak holder | Charges rapidly to each positive peak (through diode's low resistance); holds voltage between peaks |
| Resistor R | Discharge control | C discharges slowly through R — time constant $\tau = RC$ controls how fast it can follow envelope changes |
| DC block | Offset removal | Series capacitor removes the $A_c$ DC offset, leaving only $m(t)$ |
The time constant $\tau = RC$ must satisfy:
In DSB-SC, $s(t) = m(t)\cos(\omega_c t)$ — no carrier. When $m(t)$ goes negative, the phase of the carrier flips 180°. The envelope of the signal is $|m(t)|$, not $m(t)$. An envelope detector would output $|m(t)|$ — a full-wave rectified version of the message, which is completely wrong. This is why DSB-SC and SSB require a coherent local oscillator phase-locked to the carrier frequency.
The transmitted signal is $[A_c + m(t)]\cos(\omega_{IF}t)$. Total input power at the IF filter output:
Only the sideband power reaches the output as signal. The modulation efficiency is:
where $k_a = A/A_c$ is the modulation index ($\leq 1$ to avoid over-modulation). Even at $k_a = 1$, at most 50% of transmit power carries information.
Noise analysis is identical to DSB-SC: IF bandwidth is $2f_m$, so $N_i = 2n_0 f_m$. But SNR is degraded because part of $S_i$ is the useless carrier:
DSB and DSB-SC have the same noise floor ($N_i = 2n_0 f_m$, same IF bandwidth). The only difference is that DSB wastes at least half its transmit power on the carrier. To achieve the same SNR as DSB-SC, a DSB transmitter must use at least twice the power. This is why DSB-SC and SSB replaced DSB in all modern systems where power efficiency matters.
| DSB (AM) | DSB-SC | SSB | |
|---|---|---|---|
| Carrier transmitted | Yes | No | No |
| Sideband filter | No | No | Yes |
| IF bandwidth | $2f_m$ | $2f_m$ | $f_m$ |
| $N_i$ | $2n_0 f_m$ | $2n_0 f_m$ | $\frac{n_0 f_m}{2}$ |
| Wasted power | $A_c^2/2$ (carrier) | 0 | 0 |
| Power efficiency $\eta$ | $\leq 50\%$ | 100% | 100% |
| SNR$_{out}$ / SNR$_{in}$ | $< 1$ (worse) | 2 (+3 dB) | 1 (0 dB) |
| Demodulation | Envelope detector | Coherent LO | Coherent LO |
| Receiver complexity | Simplest | Medium | Most complex |
| Historical use | AM broadcast radio | Data/audio links | HF voice, satellite |
PSK = Phase Shift Keying — information is encoded in the phase of the carrier, not its amplitude or frequency. Binary means only two possible phases, one for bit 0 and one for bit 1:
| Bit | Signal | Phase |
|---|---|---|
| 1 | $+v_0\cos(\omega_c t)$ | 0° |
| 0 | $-v_0\cos(\omega_c t)$ | 180° |
The two signals are simply negatives of each other — same amplitude, same frequency, opposite phase. Written compactly: $s(t) = \pm v_0\cos(\omega_c t)$.
Amplitude is easily distorted by channel fading and attenuation. Phase is more robust — the receiver only needs to detect which of two phase states the signal is in, not measure an exact amplitude. BPSK achieves the best BER performance of all binary modulation schemes for a given $E_b/n_0$ — it is the most power-efficient binary modulation.
The receiver multiplies the received signal by a local carrier $\cos(\omega_c t)$ (coherent detection) and integrates over one bit period $T_b$:
For bit "1", $r(t) = v_0\cos(\omega_c t)$. Multiplying by the local carrier and integrating:
Start from the cosine double angle formula:
$$\cos(2\theta) = \cos^2(\theta) - \sin^2(\theta)$$Substitute $\sin^2(\theta) = 1 - \cos^2(\theta)$:
$$\cos(2\theta) = \cos^2(\theta) - (1-\cos^2(\theta)) = 2\cos^2(\theta) - 1$$Rearrange to isolate $\cos^2(\theta)$:
$$\boxed{\cos^2(\theta) = \frac{1+\cos(2\theta)}{2}}$$This is a pure algebraic rearrangement — no approximation. Replace $\theta = \omega_c t$ and you get the substitution used in the integral above.
The identity $\cos^2(\theta) = \frac{1}{2} + \frac{1}{2}\cos(2\theta)$ splits the integrand into two terms. The $\cos(2\omega_c t)$ term oscillates at twice the carrier frequency and averages to zero over any whole number of carrier cycles. Only the constant $\frac{1}{2}$ survives — that is where the $\div 2$ comes from. The interactive plot below shows both terms:
Increase the number of cycles — the red term always cancels to zero. The surviving ½ (green dashed) gives the Tb/2 result.
The result is compared against a threshold at zero. An error occurs when noise pushes the result across zero to the wrong side — this probability is the BER.
For binary PSK with threshold detection, the received signal is $r(t) = s(t) + v_n(t)$, where $s(t) = +v_0$ for a "1" and $s(t) = -v_0$ for a "0". The receiver compares the received voltage against a threshold (at zero) to decide which bit was sent. An error occurs when noise pushes the signal past the threshold:
The noisy received signal fluctuates around $+v_0$ during a "1" bit and around $-v_0$ during a "0" bit. If noise exceeds $v_0$ in magnitude, the signal crosses the threshold and causes a bit error.
Signal power $S_o$: after the coherent detector the signal output is a DC level $\pm v_0$. The power of a DC level across $1\,\Omega$ is simply:
$$S_o = v_0^2$$No factor of $\frac{1}{2}$ — that only appears for sinusoids. A DC level has the same instantaneous and average power.
Noise power $N_o$: the output noise is Gaussian with zero mean and variance $\sigma^2$. Since $E[v_n] = 0$:
$$N_o = E[v_n^2] = \sigma^2$$Therefore:
$$\frac{S_o}{N_o} = \frac{v_0^2}{\sigma^2} \qquad \Rightarrow \qquad \sqrt{\frac{S_o}{N_o}} = \frac{v_0}{\sigma}$$This ratio is exactly "how many noise standard deviations the signal sits away from the threshold" — the deeper this is into the Gaussian tail, the lower the BER.
The complementary error function is the tail area of a Gaussian curve from $x$ to $\infty$:
It appears in BER because noise is Gaussian — the probability of an error is exactly the tail area of the noise distribution beyond the decision threshold. A larger argument $x$ means the threshold is further into the tail → smaller area → lower BER.
想像你在射飛鏢Imagine you are throwing dartsダーツを投げていると想像してください,靶心是「0」這條判斷線。你丟出的飛鏢本來應該落在 and the line marked “0” is the bullseye. Your dart should land at 。「0」と書かれた線が的の中心です。投げたダーツは本来 +v₀ 的位置(代表 bit "1"),但 (representing bit "1"), but の位置(ビット "1" を表す)に落ちるはずですが、噪音noise雑音會讓飛鏢隨機偏移。 knocks it off target at random.がダーツをランダムにずらします。
噪音是高斯分布The noise is Gaussian雑音はガウス分布に従う,意思是:大部分時候偏移很小,偶爾才偏很多——就像鐘形曲線,中間高、兩邊低:, which means: most of the time the deviation is small, and only occasionally is it large — like a bell curve, high in the middle and low at the edges:、つまり、ほとんどの場合ずれは小さく、ときどきだけ大きくずれます — 中央が高く両端が低い釣鐘型の曲線です:
什麼時候出錯?When does an error occur?誤りはいつ起こるのか? 噪音把 +v₀ 的訊號往左推,推過了 0 這條線 → 接收器誤判為 "0" → 出錯。那個 The noise pushes the +v₀ signal to the left, past the 0 line → the receiver decides it was a "0" → an error. That 雑音が +v₀ の信号を左へ押しやり、0 の線を越えると → 受信機は "0" と判定します → 誤りです。その跑到門檻左邊的面積area that spills past the thresholdしきい値を越えてはみ出した面積就是「尾巴(tail)」,也就是出錯的機率: is the “tail”, and it is exactly the probability of error:が「裾(tail)」であり、まさに誤り確率です:
erfc 的括號裡是 $\sqrt{S_o/N_o}$Inside the erfc bracket is $\sqrt{S_o/N_o}$erfc の括弧の中は $\sqrt{S_o/N_o}$,就是訊號強度除以噪音強度。這個數字越大,代表訊號比噪音強很多,訊號中心離門檻很遠,鐘形曲線幾乎沒有尾巴跑到門檻那邊:, which is just the signal strength divided by the noise strength. The larger this number, the further the signal sits above the noise, the further the centre of the distribution is from the threshold, and the less of the bell curve's tail spills across it:、すなわち信号強度を雑音強度で割った値です。この値が大きいほど信号は雑音よりずっと強く、分布の中心はしきい値から遠ざかり、釣鐘型の裾がしきい値を越えてはみ出す量はほとんどなくなります:
| 情況Caseケース | 鐘形曲線Bell curve釣鐘型曲線 | 尾巴大小Size of the tail裾の大きさ | 結果Result結果 |
|---|---|---|---|
| SNR 小(訊號弱)Small SNR (weak signal)SNR 小(信号が弱い) | 中心靠近門檻,分布又寬The centre is near the threshold and the distribution is wide中心がしきい値に近く、分布も広い | 大尾巴Big tail大きな裾 | 常常出錯Errors are frequent誤りが頻発 |
| SNR 大(訊號強)Large SNR (strong signal)SNR 大(信号が強い) | 中心遠離門檻,分布又窄The centre is far from the threshold and the distribution is narrow中心がしきい値から遠く、分布も狭い | 幾乎沒有尾巴Almost no tail裾がほとんどない | 幾乎不出錯Errors are almost never madeほとんど誤らない |
一句話:In one sentence:一言でいえば: erfc 就是在算「鐘形曲線有多少面積跑到了危險區」。訊號越強、噪音越小,危險區的面積就越小,錯誤就越少。 erfc simply computes “how much of the bell curve's area has spilled into the danger zone”. The stronger the signal and the smaller the noise, the smaller that area and the fewer the errors. erfc は「釣鐘型曲線の面積のうち、どれだけが危険領域にはみ出したか」を計算しているだけです。信号が強く雑音が小さいほど、その面積は小さくなり、誤りも減ります。
送 "0" 時,訊號在 $-v_0$,整個圖左右翻轉。出錯條件變成噪音把訊號When a "0" is sent the signal sits at $-v_0$ and the whole picture is mirrored left-to-right. An error now occurs when the noise pushes the signal "0" を送るとき信号は $-v_0$ にあり、図全体が左右反転します。誤りが起こるのは、雑音が信号を往右推過 0to the right, past 0右へ押しやって 0 を越えさせた,誤判為 "1":, so it is misread as a "1":とき、つまり "1" と誤判定されるときです:
高斯分布是A Gaussian distribution is ガウス分布は對稱的symmetric対称,所以:, so:なので:
$$P(\text{error}|\text{sent "1"}) = P(n < -v_0) = P(n > v_0) = P(\text{error}|\text{sent "0"})$$不管送 "0" 還是 "1",出錯機率完全相同。這就是為什麼 BER 公式只需要寫一個——the probability of error is exactly the same whether a "0" or a "1" was sent. That is why only one BER formula is needed —"0" を送っても "1" を送っても誤り確率はまったく同じです。だから BER の式は 1 本で足ります —高斯的對稱性保證它們相等the symmetry of the Gaussian guarantees the two are equalガウス分布の対称性が両者の一致を保証する,不需要分兩種情況討論。, so there is no need to treat the two cases separately.ので、2 つの場合に分けて論じる必要はありません。
直覺上很合理——把門檻移到 $+v_0$ 旁邊,送 "1" 的那側就幾乎不會錯。但這有一個致命的代價:It sounds reasonable — move the threshold next to $+v_0$ and the "1" side will hardly ever be wrong. But there is a fatal cost:もっともらしく聞こえます — しきい値を $+v_0$ のそばへ動かせば "1" 側はほとんど誤らなくなります。しかしそこには致命的な代償があります:
把門檻往左移,$+v_0$ 那側的尾巴確實縮小了——但門檻同時往 $-v_0$ 靠近,$-v_0$ 那側的尾巴Moving the threshold left really does shrink the tail on the $+v_0$ side — but it simultaneously moves closer to $-v_0$, so the tail on that side しきい値を左へ動かせば $+v_0$ 側の裾は確かに小さくなります — しかし同時に $-v_0$ に近づくため、そちら側の裾は等量暴增grows by exactly as muchまったく同じだけ増加します。總 BER 不變:. The total BER is unchanged:。合計の BER は変わりません:
$$P_e^{total} = \frac{1}{2}P_e(\text{"1"}) + \frac{1}{2}P_e(\text{"0"})$$對稱情況下,任何偏離 0 的門檻都只是把錯誤從一側搬到另一側,In the symmetric case, any threshold away from 0 merely shifts errors from one side to the other; 対称な場合、0 から外れたしきい値は誤りを一方の側からもう一方へ移すだけであり、門檻在 0 是數學上最優的決策點a threshold at 0 is the mathematically optimal decision pointしきい値 0 が数学的に最適な判定点です。.。
縮小尾巴的正確方法不是移門檻,而是The right way to shrink the tails is not to move the threshold but to 裾を小さくする正しい方法はしきい値を動かすことではなく、增加 $+v_0$ 和 $-v_0$ 之間的間距increase the separation between $+v_0$ and $-v_0$$+v_0$ と $-v_0$ の間隔を広げる,也就是提高訊號功率。這樣兩個分布同時遠離門檻,兩側的尾巴都縮小:, that is, to raise the signal power. Both distributions then move away from the threshold at once and both tails shrink:こと、つまり信号電力を上げることです。そうすれば両方の分布が同時にしきい値から遠ざかり、両側の裾がともに小さくなります:
$E_b/n_0$ 衡量的正是「兩個訊號分布離門檻有多遠」相對於「噪音分布有多寬」。$E_b/n_0$ 越大,兩個鐘形曲線離得越遠、重疊越少、BER 越低——這是提升通訊可靠度的$E_b/n_0$ measures precisely “how far the two signal distributions are from the threshold” relative to “how wide the noise distribution is”. The larger $E_b/n_0$, the further apart the two bell curves, the less they overlap and the lower the BER — this is the $E_b/n_0$ は、まさに「2 つの信号分布がしきい値からどれだけ離れているか」を「雑音分布がどれだけ広いか」に対して測った量です。$E_b/n_0$ が大きいほど 2 つの釣鐘型曲線は離れ、重なりは減り、BER は下がります — これが通信の信頼性を高める根本方法fundamental way根本的な方法,而不是調整門檻位置。 to improve the reliability of a link, not adjusting where the threshold sits.であり、しきい値の位置を調整することではありません。
In BPSK, an error occurs when Gaussian noise $v_n$ exceeds the signal level $v_0$ and flips the decision. The probability of this is:
$$P_e = P(v_n > v_0) = \int_{v_0}^{\infty} \frac{1}{\sqrt{2\pi}\,\sigma}e^{-v^2/2\sigma^2}\,dv$$Substituting $t = v/\sigma$ converts this exactly into $\frac{1}{2}\mathrm{erfc}(v_0/\sigma\sqrt{2})$. The argument $\sqrt{S_o/N_o}$ is precisely this ratio — signal level over noise standard deviation.
| $x$ | $\mathrm{erfc}(x)$ | $P_e = \frac{1}{2}\mathrm{erfc}(x)$ | Note |
|---|---|---|---|
| 0 | 1 | 0.5 | 50% error — no SNR at all |
| 1 | 0.1573 | 0.079 | |
| 2 | 0.00468 | $2.3\times10^{-3}$ | |
| 3 | $2.2\times10^{-5}$ | $1.1\times10^{-5}$ | $\approx P_e = 10^{-5}$ target |
| 3.12 | $\approx 2\times10^{-5}$ | $10^{-5}$ | Requires $E_b/n_0 = 9.6$ dB |
| 4 | $1.5\times10^{-8}$ | $7.5\times10^{-9}$ | Very reliable link |
The two expressions in the equation above are related by:
| Term | Meaning | Units |
|---|---|---|
| $S_o/N_o$ | Output SNR at the detector | dimensionless |
| $E_b/n_0$ | Energy per bit / noise density — fundamental link quality | dimensionless |
| $R_b/B$ | Spectral efficiency — bits per second per Hz of bandwidth | bps/Hz |
Higher SNR → larger erfc argument → smaller tail → lower $P_e$. The erfc function drops very steeply — a few dB improvement in $E_b/n_0$ near the $10^{-5}$ target changes $P_e$ by orders of magnitude.
$E_b = S_o/R_b$ = energy per bit (J); $n_0 = N_o/B$ = noise PSD (W/Hz)
$E_b/n_0$ = bit energy to noise density ratio — fundamental link quality metric
Spectral efficiency: $\eta = R_b/B$ (bps/Hz). Then $S_o/N_o = (E_b/n_0)\cdot(R_b/B)$.
Step 1 — The received signal. After the BPSK detector, the received value is signal plus Gaussian noise:
Step 2 — Probability of error. For "1" sent, error occurs when $r < 0$, i.e. when noise $n < -v_0$. By symmetry of the Gaussian this equals the right tail $P(n > v_0)$:
Step 3 — Substitute $t = n/\sigma$, then $u = t/\sqrt{2}$.
Step 4 — Express in terms of $S_o/N_o$. Signal power $S_o = v_0^2$ and noise power $N_o = \sigma^2$, so:
Step 5 — Express in terms of $E_b/n_0$. By definition $E_b = S_o/R_b$ and $n_0 = N_o/B$, so:
$\underbrace{P(\text{noise crosses threshold})}_{\text{Gaussian tail integral}}$ $\xrightarrow{\;t=n/\sigma,\;u=t/\sqrt{2}\;}$ $\dfrac{1}{2}\mathrm{erfc}\!\left(\dfrac{v_0}{\sigma\sqrt{2}}\right)$ $\xrightarrow{\;S_o=v_0^2,\;N_o=\sigma^2\;}$ $\dfrac{1}{2}\mathrm{erfc}\!\left(\sqrt{\dfrac{S_o}{N_o}}\right)$ $\xrightarrow{\;E_b,\,n_0\;\text{defs}\;}$ $\dfrac{1}{2}\mathrm{erfc}\!\left(\sqrt{\dfrac{E_b}{n_0}\cdot\dfrac{R_b}{B}}\right)$
The "tail" is the probability that noise is large enough to push the signal past the decision threshold — causing a wrong bit decision. Visually, it is the shaded area under the noise distribution curve beyond the threshold:
| Tail size | Physical meaning |
|---|---|
| Large tail | Noise is strong relative to signal — frequently pushes past threshold → many bit errors |
| Small tail | Noise is weak relative to signal — rarely crosses threshold → very few errors |
Two things shrink the tail — both increase the erfc argument $\sqrt{S_o/N_o}$, moving the threshold deeper into the distribution:
① More signal power — the distribution center $+v_0$ moves further from the threshold at zero. The curve shape stays the same width, but the whole bell shifts right — so less of it spills past the threshold:
② Less noise — the distribution stays centred at the same $+v_0$, but the bell becomes narrower. Even though the center-to-threshold distance is unchanged, the narrower shape puts far less probability in the tail region:
When you double signal power, the amplitude $v_0$ grows as $\sqrt{2}$ — the distribution shifts right by $\sqrt{2}$. But when noise power doubles, $\sigma$ also grows as $\sqrt{2}$ — the bell gets wider by $\sqrt{2}$. The key asymmetry: signal shifting the center is a linear move of the whole bell, while noise widening is a spread that directly inflates the tail.
$S_o/N_o = v_0^2/\sigma^2$ captures exactly this asymmetry. The erfc argument is $\sqrt{S_o/N_o} = v_0/\sigma$:
Both effects enter through the ratio $v_0/\sigma$. The SNR is simply asking: "how many noise standard deviations away is the signal from the threshold?"
Because the Gaussian tail falls off as $e^{-x^2}$ (faster than exponential), a small increase in SNR causes a dramatic collapse in BER:
| $E_b/n_0$ | $P_e$ |
|---|---|
| 8 dB | $\sim 10^{-3}$ |
| 9.6 dB | $10^{-5}$ |
| 12 dB | $\sim 10^{-9}$ |
Just 3–4 dB of SNR improvement changes the error rate by 6 orders of magnitude. This cliff-like behaviour is why communication systems are designed to operate just above the BER target threshold — adding more power beyond that gives diminishing returns compared to the rapid gains near the knee.
Problem: A BPSK system has $E_b/n_0 = 12$ dB, $R_b = 100$ kbps, $B = 100$ kHz. Find the output SNR and estimate $P_e$.
$S_o/N_o = E_b/n_0 \cdot R_b/B = 12\text{ dB} + 10\log(100\text{k}/100\text{k}) = 12 + 0 = 12$ dB $= 15.85$ (linear)
$P_e = \frac{1}{2}\,\mathrm{erfc}(\sqrt{15.85}) = \frac{1}{2}\,\mathrm{erfc}(3.98) \approx \frac{1}{2}\times 1.1\times10^{-8} \approx 5.5\times10^{-9}$
(Very low error rate — well below the typical $10^{-5}$ requirement.)
$T_b$ = bit period — the duration of one bit, equal to the reciprocal of the bit rate:
$$T_b = \frac{1}{R_b}$$For example, $R_b = 1$ Mbps → $T_b = 1\;\mu\text{s}$ per bit. The integrate-and-dump integrates exactly over one $T_b$, then resets. Longer $T_b$ (slower bit rate) means more integration time, better SNR, and lower BER — the fundamental tradeoff between data rate and reliability.
In PSK demodulators, the integrate-and-dump detector dramatically improves BER by accumulating the signal over one bit period before making a decision. The block diagram and waveforms:
Left: Input signal $s(t)$ — binary PSK with bits 1, 1, 0, 1. Right: Integrator output $s_o(t)$ — ramps to $+VT$ for "1" and $-VT$ for "0", then resets (dumps) each bit period.
The integrator accumulates the signal over one bit period $T$, then the sampler reads the value at $t_s = N\cdot T$ and resets. The signal accumulates to $\pm VT$, while noise only grows as $\sqrt{T}$ (standard deviation), giving a much better SNR at the sample instant:
Signal is constant $\pm v_0$ every moment — every infinitesimal slice $dt$ adds in the same direction, so contributions stack linearly: $\int_0^T v_0\,dt = v_0 T$. Noise is random — each slice has a random sign and cancels partially with the next. This is a random walk: variance adds linearly, but standard deviation (amplitude) only grows as $\sqrt{T}$. The interactive plot below shows both curves side by side, and the resulting SNR improvement at the bottom:
SNR improves as √Tb — the longer the integration, the better the detection.
x-axis: t / Tb = normalized time — 0 is the start of the bit period, 1 is the end (when the sampler reads and resets). The chart shows one full bit period regardless of the actual value of Tb.
Step 1 — Error condition for bit "0". When bit "0" is sent, the integrator output signal is $s_o(T) = -VT$. An error occurs when the output noise $v_n^{out}$ pushes the total above zero:
Step 2 — Write as a Gaussian tail integral. The output noise is Gaussian with zero mean and variance $\sigma_{out}^2$:
Step 3 — Convert to erfc. Apply the substitution $u = v_n^{out}/(\sigma_{out}\sqrt{2})$. Three things change simultaneously:
| What changes | How |
|---|---|
| Variable | $v_n^{out} = \sigma_{out}\sqrt{2}\,u$ |
| Differential | $dv_n^{out} = \sigma_{out}\sqrt{2}\,du$ |
| Lower limit | $v_n^{out} = VT \;\Rightarrow\; u = \dfrac{VT}{\sigma_{out}\sqrt{2}}$ |
| Upper limit | $\infty \;\Rightarrow\; \infty$ (unchanged) |
In any definite integral substitution, the limits must transform the same way as the variable. If $u = g(v)$, then every limit $v = a$ becomes $u = g(a)$:
$$\int_{v=a}^{v=b} f(v)\,dv \xrightarrow{u=g(v)} \int_{u=g(a)}^{u=g(b)} \tilde{f}(u)\,du$$Here the lower limit is $v_n^{out} = VT$. Plug it into the substitution formula $u = v_n^{out}/(\sigma_{out}\sqrt{2})$:
$$u_{\min} = \frac{VT}{\sigma_{out}\sqrt{2}}$$That's the new lower limit — just the old value plugged in.
Now transform the exponent and the prefactor × differential separately:
Exponent: $\dfrac{(v_n^{out})^2}{2\sigma_{out}^2} = \dfrac{(\sigma_{out}\sqrt{2}\,u)^2}{2\sigma_{out}^2} = \dfrac{2\sigma_{out}^2 u^2}{2\sigma_{out}^2} = u^2$ ✓
Prefactor × differential:
Assembled:
Comparing with the definition $\mathrm{erfc}(x) = \dfrac{2}{\sqrt{\pi}}\displaystyle\int_x^\infty e^{-u^2}\,du$, we see that $\dfrac{1}{\sqrt{\pi}}\displaystyle\int_x^\infty e^{-u^2}\,du = \dfrac{1}{2}\,\mathrm{erfc}(x)$, so:
Step 4 — Substitute $E_b$ and $\sigma_{out}^2$. Two substitutions from the physics of the integrator:
The transmitted BPSK signal is $v(t) = V\cos(\omega_c t)$, a sinusoid of amplitude $V$. The average power of a sinusoid is $V^2/2$ (the ½ comes from $\langle\cos^2\rangle = \frac{1}{2}$). Energy per bit = power × bit duration $T$:
$$E_b = \frac{V^2}{2} \cdot T = \frac{V^2 T}{2}$$The factor of ½ is entirely due to the sinusoidal carrier — a DC level $V$ would give $E_b = V^2T$ with no ½.
The integrator computes $y = \frac{1}{T}\int_0^T n(t)\,dt$ (normalized by $T$). The variance of this is:
$$\sigma_{out}^2 = \frac{1}{T^2}E\!\left[\left(\int_0^T n(t)\,dt\right)^2\right] = \frac{1}{T^2}\int_0^T\int_0^T E[n(t_1)n(t_2)]\,dt_1\,dt_2$$For white noise, $E[n(t_1)n(t_2)] = n_0\,\delta(t_1-t_2)$ (autocorrelation is a delta function). The double integral collapses:
$$= \frac{1}{T^2}\int_0^T n_0\,dt = \frac{n_0 T}{T^2} = \frac{n_0}{T}$$Physically: the integrator averages noise over $T$ — longer integration means more cancellation of random fluctuations, so variance falls as $1/T$.
| Substitution | Origin |
|---|---|
| $E_b = V^2T/2$ | Sinusoidal carrier: average power = $V^2/2$, times bit period $T$ |
| $\sigma_{out}^2 = n_0/T$ | Normalized integrator: noise variance falls as $1/T$ due to averaging |
Plugging in:
The slide uses a normalized integrator output where the integration result is divided by $T$, giving a per-unit-time output. Under this convention the signal output is $V$ (not $VT$) and $\sigma_{out}^2 = n_0/T$, so:
$$\left(\frac{V}{\sqrt{2}\,\sigma_{out}}\right)^2 = \frac{V^2}{2n_0/T} = \frac{V^2 T}{2n_0} = \frac{E_b}{n_0}$$The $T^2$ cancels cleanly. The final result is the same regardless of normalization convention:
$$\boxed{P_e = \frac{1}{2}\,\mathrm{erfc}\!\left(\sqrt{\frac{E_b}{n_0}}\right)}$$$P(\text{noise} > VT)$ $\;\xrightarrow{\text{Gaussian tail}}\;$ $\dfrac{1}{2}\mathrm{erfc}\!\left(\dfrac{VT}{\sqrt{2}\,\sigma_{out}}\right)$ $\;\xrightarrow{E_b = V^2T/2,\;\sigma_{out}^2 = n_0/T}\;$ $\dfrac{1}{2}\mathrm{erfc}\!\left(\sqrt{\dfrac{E_b}{n_0}}\right)$
Signal accumulates to $VT$, noise variance is $n_0/T$ — their ratio gives $E_b/n_0$, the fundamental figure of merit.
Problem: A PSK system has $V = 0.1$ V, $T = 10\;\mu$s, $n_0 = 4\times10^{-18}$ W/Hz. Find $E_b/n_0$ and $P_e$ with the integrate-and-dump detector.
$E_b = V^2T/2 = (0.1)^2 \times 10\times10^{-6}/2 = 5\times10^{-8}$ J
$E_b/n_0 = 5\times10^{-8}/(4\times10^{-18}) = 1.25\times10^{10}$ → $10\log(1.25\times10^{10}) = 101$ dB
$P_e = \frac{1}{2}\,\mathrm{erfc}(\sqrt{1.25\times10^{10}}) \approx 0$ (essentially error-free)
BPSK (Binary PSK) — 1 bit per symbol, 2 phase states (0°, 180°): $s(t) = \pm v_0\cos(\omega_c t)$
QPSK (Quadrature PSK) — 2 bits per symbol, 4 phase states (0°, 90°, 180°, 270°), modulating both I and Q axes independently.
| BPSK | QPSK | |
|---|---|---|
| Bits per symbol | 1 | 2 |
| Phase states | 2 (0°, 180°) | 4 (0°, 90°, 180°, 270°) |
| Bandwidth | $R_b$ | $R_b/2$ — half the bandwidth for the same bit rate |
| BER formula | $\frac{1}{2}\mathrm{erfc}\!\left(\sqrt{E_b/n_0}\right)$ | $\frac{1}{2}\mathrm{erfc}\!\left(\sqrt{E_b/n_0}\right)$ |
| $E_b/n_0$ for $P_e=10^{-5}$ | 9.6 dB | 9.6 dB |
| Complexity | Simple | Needs I/Q demodulator |
QPSK is essentially two independent BPSK channels running on orthogonal I and Q axes. Each axis carries 1 bit with the same noise margin $v_0/\sigma$ as BPSK. Since the axes are orthogonal (I and Q do not interfere with each other), the BER of each is identical to BPSK — so the overall BER formula is the same.
QPSK achieves twice the spectral efficiency with no BER penalty — this is why it dominates in practice (4G/5G, satellite, cable modem).
| Modulation | $E_b/n_0$ for $P_e=10^{-5}$ | Notes |
|---|---|---|
| Coherent PSK | 9.6 dB | Best; requires sync detection |
| Coherent FSK | 12.6 dB | 3 dB worse than PSK |
| Noncoherent FSK | 13.4 dB | No carrier recovery needed |
| Coherent ASK | 15.6 dB | 6 dB worse than PSK |
| Noncoherent ASK | 16.4 dB | Simplest implementation |
Given: QPSK, $R_b = 48.6$ kbps, $B = 30$ kHz, required BER $= 10^{-5}$.
For QPSK: $P_e = \frac{1}{2}\mathrm{erfc}(\sqrt{E_b/n_0}) = 10^{-5}$ → $E_b/n_0 = 9.6$ dB
$\frac{S_o}{N_o} = \frac{E_b}{n_0}\cdot\frac{R_b}{B} = 9.6 + 10\log\!\left(\frac{48.6}{30}\right) = 9.6 + 2.1 = 11.7$ dB
Problem: A 16-QAM system has $R_b = 200$ kbps and $B = 50$ kHz. If the required $E_b/n_0$ for the target BER is 14.4 dB (from Table 14.1 in Pozar), find the minimum detectable output SNR.
$\frac{S_o}{N_o} = \frac{E_b}{n_0}\cdot\frac{R_b}{B} = 14.4 + 10\log\!\left(\frac{200\text{k}}{50\text{k}}\right) = 14.4 + 6.02 = 20.4$ dB
16-QAM has spectral efficiency $R_b/B = 4$ bps/Hz (since each symbol carries 4 bits), which increases the required SNR by 6 dB compared to $E_b/n_0$.
16-QAM carries 4 bits per symbol using both amplitude and phase — it is not enough to just demodulate I and Q like QPSK.
16-QAM uses a 4×4 constellation — 4 amplitude levels on each of I and Q axes, giving $4\times4 = 16$ points total. Each axis independently carries 2 bits (not 1 like QPSK), using 4 amplitude levels: $\pm1, \pm3$.
Q
+3 • • • •
+1 • • • •
-1 • • • •
-3 • • • •
-3 -1 +1 +3 → I
Each • = one unique 4-bit symbol
Yes and no. The I/Q demodulation step is the same as QPSK — multiply by local carrier, integrate. That part is sufficient to extract the I and Q values. But the decision step is harder:
| BPSK | QPSK | 16-QAM | |
|---|---|---|---|
| Axes | 1 | 2 (I, Q) | 2 (I, Q) |
| Levels per axis | 2 | 2 | 4 |
| Thresholds per axis | 1 | 1 | 3 |
| Bits per symbol | 1 | 2 | 4 |
In QPSK each axis only needs to decide $+$ or $-$ (one threshold at 0). In 16-QAM each axis must decide among 4 levels — requiring 3 thresholds and an amplitude detector, not just a sign detector.
The constellation points are closer together relative to their power. The minimum distance between adjacent points is smaller, so noise is more likely to push a point to a wrong neighbor.
This is why 16-QAM requires about 4 dB more $E_b/n_0$ than QPSK for the same BER — you pay an SNR penalty for the spectral efficiency gain.
On top of that, converting from $E_b/n_0$ to output SNR adds another 6 dB because of the spectral efficiency $R_b/B = 4$ bps/Hz:
So 16-QAM pays twice: ~4 dB more $E_b/n_0$ for the denser constellation, plus 6 dB from packing 4 bits into each symbol period — a total of ~10 dB more output SNR than QPSK.
Problem: An amplifier with gain $G = 15$ dB and bandwidth $B = 500$ MHz produces an output noise power of $N_o = -65$ dBm when its input is terminated with a 0 K source. Find $T_e$.
Why 0 K source? At 0 K the thermal noise from the source is zero (since thermal noise $= kTB$, and $T=0$ gives 0). This means all the output noise $N_o = -65$ dBm comes purely from the amplifier's internal noise, not from the input. This isolates the amplifier's own noise contribution, making it possible to directly solve for $T_e$.
$G = 10^{1.5} = 31.62$; $N_o = 10^{(-65-30)/10} = 3.16\times10^{-10}$ W
$T_e = \frac{N_o}{GkB} = \frac{3.16\times10^{-10}}{31.62\times1.38\times10^{-23}\times500\times10^6} = \frac{3.16\times10^{-10}}{2.18\times10^{-13}} = 1449$ K
Problem: An amplifier with gain $G = 10$ dB and bandwidth $B = 100$ MHz produces an output noise power of $N_o = -80$ dBm when its input is terminated with a source at $T_s = 290$ K. Find $T_e$.
Key difference from Example 7: Since the source is not at 0 K, the output noise now contains both amplified source thermal noise and the amplifier's own noise: $N_o = G \cdot k \cdot (T_s + T_e) \cdot B$. We must subtract the source contribution to isolate $T_e$.
$G = 10^{10/10} = 10$; $N_o = 10^{(-80-30)/10} = 10^{-11}$ W
$T_e = \frac{N_o}{GkB} - T_s = \frac{10^{-11}}{10\times1.38\times10^{-23}\times100\times10^6} - 290 = \frac{10^{-11}}{1.38\times10^{-14}} - 290 = 724.6 - 290 = 435$ K
Given: X-band amp, $G=20$ dB, $B=1$ GHz. $T_1=290$ K → $N_1=-62$ dBm; $T_2=77$ K → $N_2=-64.7$ dBm.
$Y = 10^{2.7/10} = 1.86$
$T_e = (290 - 1.86\times77)/(1.86-1) = (290-143.2)/0.86 = 170$ K
If a source with an equivalent noise temperature of $T_s = 450$ K drives the amplifier with a gain of 20 dB (=100), the source into the amplifier is $kT_sB$. The total noise power out of the amplifier can be calculated as:
$N_o = G\,k\,B(T_s+T_e) = 100\times1.38\times10^{-23}\times10^9\times620 = 8.56\times10^{-10}$ W $= -60.7$ dBm
Here $10^9$ comes from $B = 1\text{ GHz} = 1\times10^9\text{ Hz}$, converting to base SI units.
So whenever you see $(F-1)\times 290$, the 290 is not an approximation — it is the defined constant $T_0$.
What $F$ means: The noise figure $F$ is defined as the SNR degradation factor: $F = SNR_{in}/SNR_{out}$. "Noise figure" and "SNR degradation" are the same concept — $F$ measures how much the device worsens the signal-to-noise ratio.
Naming convention: Some textbooks call the linear ratio $F$ the noise factor and reserve noise figure for the dB value. Others (including Pozar) use "noise figure" for both and let context distinguish them.
Why $NF = 10\log_{10}(F)$? Because $F$ is a power ratio ($SNR_{in}/SNR_{out}$, where each SNR is signal power / noise power). The dB scale for power ratios is always $10\log_{10}$ — same as gain ($G_{\text{dB}} = 10\log_{10}G$) or loss ($L_{\text{dB}} = 10\log_{10}L$). For voltage ratios it would be $20\log_{10}$ instead, because power $\propto V^2$.
From the definition $F = \frac{S_i/N_i}{S_o/N_o}$, substitute $S_o = G\cdot S_i$ and $N_o = G\cdot kB(T_0 + T_e)$:
$$F = \frac{S_i/(kT_0B)}{G\cdot S_i/[G\cdot kB(T_0+T_e)]} = \frac{T_0+T_e}{T_0} = 1 + \frac{T_e}{T_0}$$The gain $G$ and signal $S_i$ cancel — $F$ depends only on how much extra noise $T_e$ the network adds relative to $T_0 = 290$ K.
Rearranging:
$$F - 1 = \frac{T_e}{T_0} \;\;\Longrightarrow\;\; T_e = (F-1)\cdot 290\text{ K}$$Problem: A low-noise amplifier is tested with $T_1 = 290$ K ($N_1 = -70$ dBm) and $T_2 = 77$ K ($N_2 = -71.5$ dBm). Find $T_e$ and the noise figure $F$.
$Y = 10^{((-70)-(-71.5))/10} = 10^{0.15} = 1.413$
$T_e = \frac{290 - 1.413\times77}{1.413-1} = \frac{290 - 108.8}{0.413} = \frac{181.2}{0.413} = 438.7$ K
$F = 1 + T_e/T_0 = 1 + 438.7/290 = 2.513$
$\text{NF} = 10\log_{10}(F) = 10\log_{10}(2.513) = 4.0\text{ dB}$
General output noise:
$$N_o = G\,(N_i + N_{added}) \;\;\Rightarrow\;\; T_e = \frac{N_{added}}{k\,B}$$General noise figure (depends on $N_i$):
$$F = \frac{S_i / N_i}{S_o / N_o} = \frac{S_i / N_i}{G\,S_i\,/\,[G(N_i + N_{added})]} = 1 + \frac{N_{added}}{N_i}$$Since in general $N_{added}$ cannot be proportional to $N_i$, the noise figure depends on $N_i$. But in real noise figure measurement, $N_i$ is defined at $kT_0B$ with $T_0 = 290$ K, such that a fixed value of noise figure can be taken:
$$F \equiv 1 + \frac{N_{added}}{N_i(T_0)} = 1 + \frac{k\,T_e\,B}{k\,T_0\,B} = 1 + \frac{T_e}{T_0} = 1 + \frac{T_e}{290}$$Output noise with $T_e$ model:
$N_i$: input noise power; $k\,T_e\,B$: internally generated noise from amplifier.
$$N_o = G\cdot(\underbrace{N_i}_{\text{input}} + \underbrace{k\,T_e\,B}_{\text{amplifier}}) = G\cdot(N_i + N_{added}) = k\,G\,B\,(T_0 + T_e)$$Noise figure from SNR ratio:
$$F = \frac{S_i}{N_i}\cdot\frac{N_o}{S_o} = \frac{S_i}{k\,T_0\,B}\cdot\frac{k\,G\,B\,(T_0+T_e)}{G\cdot S_i} = 1 + \frac{T_e}{T_0} \;\;\Longleftrightarrow\;\; T_e = (F-1)\,T_0 \tag{2}$$Problem: An LNA has a noise figure of 0.8 dB. What is its equivalent noise temperature? If a second amplifier has $T_e = 500$ K, what is its noise figure in dB?
LNA (NF → $T_e$):
Convert dB to linear: $F = 10^{NF/10} = 10^{0.8/10} = 10^{0.08} = 1.202$
Apply Eq. (1): $T_e = (F-1)\,T_0 = (1.202-1)\times 290 = 0.202\times 290 = 58.6$ K
Second amp ($T_e$ → NF):
Rearrange Eq. (1) the other way — solve for $F$: $$F = 1 + \frac{T_e}{T_0} = 1 + \frac{500}{290} = 1 + 1.724 = 2.724$$
Convert to dB: $NF = 10\log_{10}(F) = 10\log_{10}(2.724) = 4.35$ dB
Physical meaning of $G$: A lossy transmission line with insertion loss $L$ is treated as a two-port with available gain $G = 1/L < 1$. A matched source (resistor $R$ at temperature $T$) delivers input noise $N_i = kTB$. In thermal equilibrium the output noise must also be $N_o = kTB$, because the lossy line at temperature $T$ looks like another resistor at $T$ from the output port.
The setup. The line itself sits at temperature $T$, has matched characteristic impedance $Z_0 = R$, and is fed by a source also at temperature $T$. Here $L$ is the insertion loss and $G = 1/L$ is the available gain.
The thermodynamic argument. The second law of thermodynamics states that the entropy of an isolated system always increases over time, and heat spontaneously flows from hotter to colder regions. When every part of the system is at the same temperature $T$, the 2nd law forbids any net flow of energy between the parts. If $N_o$ were greater or less than $N_i$, you could place a load at one end and extract work from a single-temperature reservoir — a perpetual motion machine of the second kind. So the noise entering and leaving must balance.
Step-by-step noise balance at the output port:
The attenuated input plus the line's self-emitted noise sum to exactly $kTB$, independent of how lossy the line is.
Due to thermodynamic equilibrium:
$$N_o = N_i = k\,T\,B = G\,(N_i + N_{added})$$ $$\Rightarrow\; N_{added} = \frac{1-G}{G}\,k\,T\,B = \left(\frac{1}{G}-1\right)k\,T\,B = (L-1)\,k\,T\,B$$Equivalent noise temperature:
$$T_e = \frac{N_{added}}{k\,B} = (L-1)\cdot T$$Noise figure:
$$F = 1 + \frac{N_{added}}{N_i(T_0)} = 1 + \frac{(L-1)\,k\,T\,B}{k\,T_0\,B} = 1 + (L-1)\frac{T}{T_0} \;\xrightarrow{T=T_0}\; F = L$$$\therefore\; F(\text{dB}) = IL(\text{dB})$ under the condition $T = T_0$, where $IL$ stands for Insertion Loss (插入損耗). (insertion loss).(挿入損失)。
Problem: A coaxial cable has 3 dB loss and is at a physical temperature of $T = 350$ K. Find $T_e$ and $F$.
$L = 10^{0.3} = 2.0$
$T_e = (2-1)\times350 = 350$ K
$F = 1 + (2-1)\times350/290 = 1 + 1.207 = 2.207 = 3.44$ dB
(Note: if $T=290$ K, $F$ would be exactly $L = 2 = 3$ dB. Since the cable is hotter, $F$ is slightly higher.)
Derivation: Friis formula for two stages
Output noise of stage 1:
$$N_1 = G_1(k\,T_0\,B + k\,T_{e1}\,B)$$Output noise of the cascade (stage 2 amplifies $N_1$ and adds its own noise):
$$N_o = G_2(N_1 + k\,T_{e2}\,B) = G_1 G_2\,k\,B\left(T_0 + T_{e1} + \frac{T_{e2}}{G_1}\right)$$ $$= G_1 G_2(k\,T_0\,B + k\,T_e\,B)$$Equivalent noise temperature of cascaded system:
$$\Rightarrow\; T_e = T_{e1} + \frac{T_{e2}}{G_1}$$Noise figure:
$$F = \frac{S_i}{N_i}\cdot\frac{N_o}{S_o} = \frac{S_i}{k\,T_0\,B}\cdot\frac{\colorbox{#f8d0d0}{$G_1 G_2$}\,k\,B\,(T_0 + T_{e1} + T_{e2}/G_1)}{\colorbox{#f8d0d0}{$G_1 G_2\,S_i$}}$$ $$= 1 + \frac{T_{e1}}{T_0} + \frac{T_{e2}}{G_1 T_0} = F_1 + \frac{1}{G_1}\left[\left(1+\frac{T_{e2}}{T_0}\right)-1\right] = F_1 + \frac{F_2-1}{G_1}$$General form for arbitrary number of cascaded stages:
$$T_e = T_{e1} + \frac{T_{e2}}{G_1} + \frac{T_{e3}}{G_1 G_2} + \cdots$$ $$F = F_1 + \frac{F_2-1}{G_1} + \frac{F_3-1}{G_1 G_2} + \cdots$$LNA: $F_1=2$ dB $=1.58$, $G_1=10$ dB $=10$. BPF: $F_2=1$ dB $=1.26$, $G_2=-1$ dB $=0.79$. Mixer: $F_3=4$ dB $=2.51$, $G_3=-3$ dB $=0.5$.
$F = 1.58 + \frac{0.26}{10} + \frac{1.51}{10\times0.79} = 1.58 + 0.026 + 0.191 = 1.80 = 2.55$ dB
From Eq. (1): $T_e = (F-1)\,T_0 = (1.80-1)\times290 = 232$ K. $G_{total} = 10\times0.79\times0.5 = 3.95 = 5.97$ dB
Problem: Consider two configurations for a receiver with an LNA ($G = 20$ dB, $F = 1.5$ dB) and a cable ($L = 2$ dB):
Config A: LNA → Cable Config B: Cable → LNA
Find the cascade noise figure for each. (Assume cable at $T_0 = 290$ K.)
LNA: $F_1 = 10^{0.15} = 1.413$, $G_1 = 100$. Cable: $F = L = 10^{0.2} = 1.585$, $G = 1/L = 0.631$.
Config A (LNA first): $F = 1.413 + \frac{1.585-1}{100} = 1.413 + 0.006 = 1.419 = 1.52$ dB
Config B (Cable first): $F = 1.585 + \frac{1.413-1}{0.631} = 1.585 + 0.654 = 2.240 = 3.50$ dB
Conclusion: Placing the cable before the LNA adds ~2 dB to the noise figure! Always put the LNA first.
A transmission line has characteristic impedance $Z_0$. When the load $Z_L = Z_0$, all power is absorbed — no reflection. Drag the sliders to see what happens when $Z_L \neq Z_0$.
$$\Gamma_L = \frac{Z_L - Z_0}{Z_L + Z_0}$$
S-parameters describe how an RF signal scatters at a two-port network. $L$ is the loss factor: how much power the line absorbs as heat.
| Parameter | Meaning | Value |
|---|---|---|
| $S_{21}$ | Forward transmission (signal passes through) | 0.707 |
| $S_{11}$ | Input reflection (bounces back at port 1) | 0.000 |
| $S_{22}$ | Output reflection (bounces back at port 2) | = 0 (matched load) |
| $S_{12}$ | Reverse transmission | 0.707 |
| $L$ | Loss factor = $1/|S_{21}|^2$ | 2.000 |
Passive networks are circuits built only from passive components (resistors, capacitors, inductors, transmission lines, cables, attenuators, filters, etc.) with no gain ($G \leq 1$). The only noise they produce is thermal noise (Johnson-Nyquist noise) from random thermal motion of electrons, characterized by $N = kTB$.
Active devices (diodes, transistors) generate additional non-thermal noises: shot noise (discrete carrier crossings at junctions), flicker/1/$f$ noise (carrier trapping at surface defects), and generation-recombination noise.
Because a lossy line is purely passive, the thermodynamic equilibrium argument ($N_o = N_i = kTB$) holds perfectly — no extra non-thermal noise sources exist. This is why $F = L$ (at $T = T_0$) works for cables and attenuators but not for amplifiers or mixers containing active devices.
Derivation: $T_e$ and $F$ for a mismatched passive network
Output noise using available power gain:
$$N_2 = G_A(\Gamma_s)\cdot(N_1 + N_{added}) = G_A(\Gamma_s)\cdot(k\,T\,B + N_{added})$$where $G_A(\Gamma_s)$ is the available power gain, which depends on the source reflection coefficient $\Gamma_s$.
Thermodynamic equilibrium requires $N_2 = k\,T\,B$, therefore:
$$\Rightarrow\; N_{added} = \frac{1 - G_A(\Gamma_s)}{G_A(\Gamma_s)}\,k\,T\,B$$Equivalent noise temperature:
$$\Rightarrow\; T_e = \frac{N_{added}}{k\,B} = \frac{1 - G_A(\Gamma_s)}{G_A(\Gamma_s)}\cdot T$$Noise figure:
$$\Rightarrow\; F = 1 + \frac{N_{added}}{N_i(T_0)} = 1 + \frac{1 - G_A(\Gamma_s)}{G_A(\Gamma_s)}\,\frac{k\,T\,B}{k\,T_0\,B} = 1 + \frac{1 - G_A(\Gamma_s)}{G_A(\Gamma_s)}\cdot\frac{T}{T_0}$$The propagation constant $\gamma$ describes how a wave changes as it travels along a transmission line. It has two parts:
| Symbol | Name | Units | What it controls |
|---|---|---|---|
| $\alpha$ | Attenuation constant | Np/m | How fast the amplitude decays — caused by conductor loss (skin effect) and dielectric loss |
| $\beta$ | Phase constant | rad/m | How fast the phase rotates — related to wavelength by $\beta = 2\pi/\lambda$ |
A voltage wave of initial amplitude $V^+$, after traveling distance $l$:
The $j$ in $j\beta$ is the imaginary unit ($j^2 = -1$), so $e^{-j\beta l}$ produces phase rotation (a circle on the complex plane) rather than amplitude change.
At low frequencies, circuits are described by voltages and currents (Z, Y, ABCD matrices). At microwave frequencies this breaks down because:
S-parameters solve this by working with traveling waves instead: the incident wave $a$ going into a port and the scattered (outgoing) wave $b$ coming out. These are easy to measure with a network analyzer and well-defined at any reference plane.
Normalized wave amplitudes at each port:
$V_i^+$ = incident voltage wave at port $i$, $\;V_i^-$ = reflected voltage wave, $\;Z_0$ = reference impedance
For a two-port network:
| Parameter | Definition | Physical meaning | Common dB name |
|---|---|---|---|
| $S_{11}$ | $b_1/a_1\big|_{a_2=0}$ | Input reflection coefficient — how much bounces back at port 1 | Return loss = $-20\log|S_{11}|$ |
| $S_{21}$ | $b_2/a_1\big|_{a_2=0}$ | Forward transmission — signal from port 1 → port 2 | Gain (or insertion loss) = $20\log|S_{21}|$ |
| $S_{12}$ | $b_1/a_2\big|_{a_1=0}$ | Reverse transmission — signal from port 2 → port 1 | Isolation = $-20\log|S_{12}|$ |
| $S_{22}$ | $b_2/a_2\big|_{a_1=0}$ | Output reflection coefficient — how much bounces back at port 2 | Output return loss = $-20\log|S_{22}|$ |
The condition "$a_2 = 0$" means port 2 is terminated with a matched load ($Z_0$) so no wave enters from that side. In practice, a network analyzer does exactly this: it sends a signal into one port and terminates the other with $50\;\Omega$.
| Property | Condition | Meaning |
|---|---|---|
| Reciprocal | $[S]^T = [S]$, i.e. $S_{ij} = S_{ji}$ | Network looks the same from both sides (passive networks without ferrites) |
| Lossless | $[S]^H[S] = [I]$ (unitary) | All power entering leaves — no dissipation. $|S_{11}|^2 + |S_{21}|^2 = 1$ |
| Matched | $S_{ii} = 0$ | No reflection at port $i$ |
| Device | $[S]$ | Notes |
|---|---|---|
| Matched lossy line | $\begin{bmatrix} 0 & e^{-\gamma l} \\ e^{-\gamma l} & 0 \end{bmatrix}$ | Reciprocal, matched. $|S_{21}|^2 = e^{-2\alpha l} = 1/L$ |
| Lossless line | $\begin{bmatrix} 0 & e^{-j\beta l} \\ e^{-j\beta l} & 0 \end{bmatrix}$ | $\alpha=0$, so $|S_{21}|=1$. Unitary. |
| Ideal amplifier (gain $G$) | $\begin{bmatrix} 0 & 0 \\ \sqrt{G} & 0 \end{bmatrix}$ | Unilateral ($S_{12}=0$), matched, not reciprocal |
| Ideal isolator | $\begin{bmatrix} 0 & 0 \\ 1 & 0 \end{bmatrix}$ | Passes forward, blocks reverse. Not reciprocal (uses ferrite). |
Diagonal = 0: Both ports see impedance $Z_0$, so $\Gamma = (Z_0 - Z_0)/(Z_0 + Z_0) = 0$ — no reflection.
Off-diagonal = $e^{-\gamma l}$: The signal propagates through with attenuation $e^{-\alpha l}$ and phase shift $e^{-j\beta l}$.
Symmetric ($S_{21} = S_{12}$): The line is reciprocal — it looks the same from both ends. This holds for all passive networks without magnetically biased materials (ferrites).
Matched means $Z_g = Z_0$ (source impedance equals the line's characteristic impedance), so $\Gamma_s = 0$ — a wave arriving at the source is fully absorbed, no bounce back.
Source mismatch means $Z_g \neq Z_0$, so $\Gamma_s \neq 0$ — the source acts like a partial mirror. When a wave hits it, part reflects back into the line.
A lossy line generates thermal noise and radiates it out of both ports. The noise going out port 2 reaches the load — that's expected. But noise also goes out port 1 toward the source:
Analogy: Shouting in a hallway. If the far wall is padded (matched), your echo dies. If the far wall is concrete (mismatched), the echo bounces back and adds to the noise in the room.
The noise figure formula shows this: $F$ increases with $|\Gamma_s|^2$ because more mismatch = more reflected noise re-entering the system.
Derivation: Mismatched lossy line via S-parameters
Source reflection coefficient (mismatch at port 1):
$$\Gamma_s = \frac{Z_g - Z_0}{Z_g + Z_0} \neq 0$$S-matrix of the lossy line (length $l$, propagation constant $\gamma = \alpha + j\beta$):
$$[S] = \begin{bmatrix} 0 & e^{-\gamma l} \\ e^{-\gamma l} & 0 \end{bmatrix} = \begin{bmatrix} 0 & 1 \\ 1 & 0 \end{bmatrix} e^{-(\alpha+j\beta)l} = \begin{bmatrix} 0 & 1 \\ 1 & 0 \end{bmatrix} \frac{e^{-j\beta l}}{e^{\alpha l}} = \begin{bmatrix} 0 & 1 \\ 1 & 0 \end{bmatrix} \frac{e^{-j\beta l}}{\sqrt{L(l)}}$$The power loss factor $L$ is defined as the ratio of input to output power. Since power $\propto |\text{amplitude}|^2$:
$$L(l) = \frac{P_{\text{in}}}{P_{\text{out}}} = \frac{|V^+|^2}{|V^+ e^{-\alpha l}|^2} = e^{2\alpha l}$$Taking the square root:
$$e^{\alpha l} = \sqrt{e^{2\alpha l}} = \sqrt{L(l)}$$Therefore the amplitude attenuation factor is:
$$e^{-\alpha l} = \frac{1}{\sqrt{L(l)}}$$General formula (any two-port):
Imagine looking into port 2. What reflection do you see?
Total: $\;\Gamma_{out} = S_{22} + \dfrac{S_{12}\,S_{21}\,\Gamma_s}{1 - S_{11}\,\Gamma_s}$
Substituting the matched lossy line ($S_{11}=0,\;S_{22}=0,\;S_{12}=S_{21}=e^{-j\beta l}/\sqrt{L}$):
A lossier line (larger $L$) makes $|\Gamma_{out}|$ smaller — the source mismatch is hidden behind the loss.
Available power gain:
$$G_A(\Gamma_s) = \frac{|S_{21}|^2(1-|\Gamma_s|^2)}{|1-S_{11}\Gamma_s|^2(1-|\Gamma_{out}(\Gamma_s)|^2)} = \bbox[#ffcccc,3px]{\frac{(1/L)(1-|\Gamma_s|^2)}{1-|\Gamma_s/L|^2}}$$Equivalent noise temperature and noise figure:
$$T_e = \frac{1-G_A(\Gamma_s)}{G_A(\Gamma_s)}\,T = \frac{(L-1)(L+|\Gamma_s|^2)}{L(1-|\Gamma_s|^2)}\,T$$ $$F = 1 + \frac{T_e}{T_0} = 1 + \frac{(L-1)(L+|\Gamma_s|^2)}{L(1-|\Gamma_s|^2)}\cdot\frac{T}{T_0}$$If $\Gamma_s = 0$: $\;\;T_e = (L-1)\cdot T$ and $F = 1 + (L-1)\dfrac{T}{T_0}$, recovering the matched case.
When $\Gamma_s = 0$, this reduces to the expressions for a matched lossy line. Compare to see that the noise figure or equivalent noise temperature increases for a mismatched lossy line. The reason for this increase is that the lossy line actually delivers noise power out of both its ports, but when the input port is mismatched, some of the available noise power at port 1 is reflected from the source back into port 1, and appears at port 2. On the contrary, when the source is matched to port 1, none of the available power from port 1 is reflected back into the line, so the noise power available at port 2 is a minimum.
"when the source is matched to port 1" — $Z_g = Z_0$, so $\Gamma_s = 0$. The source impedance equals the line's characteristic impedance.
"none of the available power from port 1 is reflected back into the line" — The lossy line radiates thermal noise out of both ports. The noise going out port 1 hits the source. Since the source is matched, it absorbs 100% of that noise — nothing bounces back.
"so the noise power available at port 2 is a minimum" — Port 2 only receives two noise contributions:
No extra term exists. This is the best case — the lowest possible noise at the output.
If the source were mismatched ($\Gamma_s \neq 0$), a third contribution would appear: the reflected noise from port 1 bouncing off the source, traveling back through the line, and arriving at port 2. That extra term makes the output noise larger than the minimum.
Therefore: matched source = lowest achievable noise figure for this lossy line. Any mismatch can only make it worse.
S-matrix of the Wilkinson divider (with loss $L$ in each arm):
$$[S] = \frac{-j}{\sqrt{2L}} \begin{bmatrix} 0 & 1 & 1 \\ 1 & 0 & 0 \\ 1 & 0 & 0 \end{bmatrix}$$Derivation: Noise figure of the Wilkinson divider
Port roles in the Wilkinson divider:
Port 1 = input (signal enters)
Port 2 = output (half power, −3 dB)
Port 3 = output (half power, −3 dB)
It's a power splitter: 1 in → 2 out. For noise figure analysis, we use only port 2 as output and terminate port 3 with $Z_0$ to ground, dumping its power into the resistor.
Port 2 vs Port 3 — why different?
Port 3: Z₀ resistor to ground — a dummy load that absorbs power (not used, signal is thrown away).
Port 2: connects to the next stage (amplifier, instrument, etc.) which also presents $Z_0$ impedance — but it's the active output where you take the signal out, not a termination to ground.
The $Z_0$ label on port 2's line = characteristic impedance of the output transmission line going to the next stage, not a grounded resistor.
Why does port 3 need a dummy load but port 2 doesn't?
S-parameter gain/noise analysis needs a 2×2 matrix — but the Wilkinson has 3 ports. One port must be eliminated.
| If port 3 is... | $\Gamma_3$ | Result |
|---|---|---|
| Open (nothing connected) | $+1$ | Total reflection — waves bounce back, corrupt S-matrix |
| Shorted to ground | $-1$ | Total reflection (inverted) — same problem |
| Terminated with $Z_0$ | $0$ | No reflection — port cleanly disappears |
Port 2 doesn't need a dummy load because it's the port you're actually using — the signal comes out there to your real circuit (amplifier, analyzer, etc.).
Analogy: a Y-splitter for a garden hose. You want water from one output only. You don't cap the other (pressure builds up = reflection). You attach a drain so water flows freely without back-pressure.
How does port 2 physically connect to the next stage?
RF circuits use transmission lines (not discrete wires). Every signal path has 2 conductors: a signal trace and a ground plane.
Port 3 is identical physically — but instead of going to a next stage, its signal trace ends at a 50Ω SMD resistor soldered to the ground pad. Dead end.
To evaluate the noise figure, we first terminate port 3 with a matched load; this converts the 3-port device to a 2-port device.
What is the "termination" at port 3?
Zigzag = resistor, value $Z_0$ (e.g. 50 Ω)
Horizontal lines = ground symbol
In a real circuit: a 50 Ω resistor soldered between the port 3 trace and the ground plane.
This absorbs all incoming power → $\Gamma_3 = \frac{Z_0 - Z_0}{Z_0 + Z_0} = 0$ → no reflection back into port 3.
The full 3-port S-matrix of the Wilkinson divider (with lossy arms) is:
$$[S]_{3\times3} = \frac{-j}{\sqrt{2L}} \begin{bmatrix} 0 & 1 & 1 \\ 1 & 0 & 0 \\ 1 & 0 & 0 \end{bmatrix}$$How does 3×3 reduce to 2×2?
The S-parameter equations for 3 ports are:
$$b_1 = S_{11}a_1 + S_{12}a_2 + S_{13}a_3$$ $$b_2 = S_{21}a_1 + S_{22}a_2 + S_{23}a_3$$ $$b_3 = S_{31}a_1 + S_{32}a_2 + S_{33}a_3$$When port 3 is terminated with a matched load ($Z_L = Z_0$), the load reflection coefficient $\Gamma_3 = 0$, so $a_3 = \Gamma_3 \cdot b_3 = 0$ — no wave reflects back into port 3.
With $a_3 = 0$, the third column vanishes from the first two equations:
$$b_1 = S_{11}a_1 + S_{12}a_2$$ $$b_2 = S_{21}a_1 + S_{22}a_2$$This is a 2×2 system — just the upper-left submatrix:
$$[S]_{2\times2} = \frac{-j}{\sqrt{2L}} \begin{bmatrix} 0 & 1 \\ 1 & 0 \end{bmatrix}$$What if $\Gamma_3 \neq 0$? Then $a_3 = \Gamma_3 \cdot b_3 \neq 0$, and the signal bounces back from port 3 into ports 1 and 2. The effective 2×2 parameters become:
$$S'_{ij} = S_{ij} + \frac{S_{i3}\,S_{3j}\,\Gamma_3}{1 - S_{33}\,\Gamma_3}, \quad i,j \in \{1,2\}$$With matched termination ($\Gamma_3=0$), all correction terms vanish and $S'_{ij} = S_{ij}$.
What if we terminate port 2 instead? Set $a_2=0$ and take rows/columns $\{1,3\}$:
$$[S']_{2\times2} = \frac{-j}{\sqrt{2L}} \begin{bmatrix} S_{11} & S_{13} \\ S_{31} & S_{33} \end{bmatrix} = \frac{-j}{\sqrt{2L}} \begin{bmatrix} 0 & 1 \\ 1 & 0 \end{bmatrix}$$Exactly the same result. Ports 2 and 3 are symmetric in the Wilkinson divider (identical arm impedance $\sqrt{2}\,Z_0$ and length $\lambda/4$), so it doesn't matter which output port you use — the 2-port S-matrix, gain, and noise figure are identical.
How $\sqrt{2}\,Z_0$ and $\lambda/4$ produce the S-matrix — step by step:
Step 1: $\lambda/4$ transformer formula
| Symbol | Name | Meaning |
|---|---|---|
| $Z_{in}$ | Input impedance | What you "see" looking into one end of the line |
| $Z_T$ | Characteristic impedance | Property of the line itself (geometry/material) |
| $Z_L$ | Load impedance | What's connected at the far end |
| $\lambda/4$ | Quarter wavelength | Length of the line = one quarter of signal wavelength |
The general input impedance of a transmission line of length $l$:
$$Z_{in} = Z_T \frac{Z_L + jZ_T\tan(\beta l)}{Z_T + jZ_L\tan(\beta l)}$$For $\lambda/4$: $\;\beta l = \frac{2\pi}{\lambda}\cdot\frac{\lambda}{4} = \frac{\pi}{2}$, so $\tan(\beta l) \to \infty$. Both numerator and denominator are dominated by the $\tan$ terms:
$$Z_{in} = Z_T \frac{\cancel{Z_L} + jZ_T\tan(\beta l)}{\cancel{Z_T} + jZ_L\tan(\beta l)} \;\approx\; Z_T \cdot\frac{jZ_T\cancel{\tan(\beta l)}}{jZ_L\cancel{\tan(\beta l)}} = \frac{Z_T^2}{Z_L}$$A $\lambda/4$ line "flips" the impedance: big load → small input, small load → big input. $Z_T$ controls the scaling.
Step 2: $\sqrt{2}\,Z_0$ gives $S_{11}=0$ (port 1 matched)
Each arm has $Z_T = \sqrt{2}\,Z_0$ and is terminated by $Z_L = Z_0$ (at port 2 or 3):
$$Z_{in,\text{arm}} = \frac{(\sqrt{2}\,Z_0)^2}{Z_0} = \frac{2Z_0^2}{Z_0} = 2Z_0$$Port 1 sees two arms in parallel:
$$Z_{in,\text{total}} = \frac{2Z_0 \;\|\; 2Z_0}{} = \frac{2Z_0}{2} = Z_0 \;\;\checkmark$$ $$\Rightarrow\; \bbox[#ffcccc,3px]{\Gamma_{in} = \frac{Z_{in}-Z_0}{Z_{in}+Z_0} = \frac{Z_0 - Z_0}{Z_0 + Z_0} = 0 \;\;\Rightarrow\;\; \boxed{S_{11}=0}}$$Step 3: $\lambda/4$ gives the $-j$ phase in $S_{21}$
A lossless $\lambda/4$ line has transmission matrix phase $e^{-j\beta l} = e^{-j\cdot(2\pi/\lambda)\cdot(\lambda/4)} = e^{-j\pi/2} = -j$.
Step 4: Combine all factors for $S_{21}$
| Factor | Origin | Value |
|---|---|---|
| Phase | $\lambda/4$ arm length | $-j$ |
| Power split | 2 equal arms → half power each → voltage $\times\frac{1}{\sqrt{2}}$ | $\frac{1}{\sqrt{2}}$ |
| Loss | Each arm has insertion loss $L$ → voltage $\times\frac{1}{\sqrt{L}}$ | $\frac{1}{\sqrt{L}}$ |
Step 5: By symmetry, $S_{22}=0$ (port 2 also matched due to the $2Z_0$ isolation resistor), $S_{12}=S_{21}$ (reciprocal network).
So $\sqrt{2}\,Z_0$ and $\lambda/4$ are fully consumed into the S-matrix — they don't appear in later formulas because their job is done: they produced $S_{11}=S_{22}=0$ and $S_{21}=-j/\sqrt{2L}$.
Interactive: see how $Z_T$ controls the match
What does $\Gamma_s = 0$ mean?
$$\Gamma_s = \frac{Z_s - Z_0}{Z_s + Z_0} = 0 \;\;\Longleftrightarrow\;\; Z_s = Z_0$$The source impedance equals the characteristic impedance — no mismatch, no reflection at the input port. All available power from the source enters the network.
Matched source: $Z_s = Z_0$ means all power flows into the network ($a_1$), with zero reflected wave ($b_1 = \Gamma_s \cdot a_1 = 0$).
| Method | How it works |
|---|---|
| Already 50Ω | Most RF signal generators & amplifier outputs are designed with $Z_{out}=50\,\Omega$ |
| λ/4 transformer | Insert a quarter-wave line with $Z_T = \sqrt{Z_s \cdot Z_0}$ between source and network |
| L / π / T network | Lumped LC elements to transform impedance at a specific frequency |
| Stub matching | Shunt/series transmission line stubs designed on a Smith chart |
| Attenuator pad | Resistive network — forces both ports toward $Z_0$ at the cost of signal loss |
Interactive: impedance matching explorer
Why $G_A(\Gamma_s=0) = |S_{21}|^2$?
The general available gain formula is:
$$G_A = \frac{|S_{21}|^2\,(1-|\Gamma_s|^2)}{|1 - S_{11}\Gamma_s|^2\,(1 - |\Gamma_{out}|^2)}$$Set $\Gamma_s = 0$ (source matched to $Z_0$), each factor simplifies:
| Term | General | $\Gamma_s=0$ |
|---|---|---|
| Numerator factor | $1-|\Gamma_s|^2$ | $1$ |
| Input mismatch | $|1-S_{11}\Gamma_s|^2$ | $1$ |
| $\Gamma_{out}$ | $S_{22}+\frac{S_{12}S_{21}\Gamma_s}{1-S_{11}\Gamma_s}$ | $S_{22}$ |
So $G_A = \dfrac{|S_{21}|^2}{1-|S_{22}|^2}$. This equals $|S_{21}|^2$ only when $S_{22}=0$ (output port matched).
Here, the terminated Wilkinson divider has $S_{11}=S_{22}=0$ (matched at both ports), so all reflection terms vanish and the gain is purely the forward transmission $|S_{21}|^2 = \frac{1}{2L}$.
$\therefore$ When $T = T_0$, $F = 2L$ $\;\Leftrightarrow\;$ $F(\text{dB}) = 3\text{ dB} + \text{Loss of the line in dB from port 1 to 2} = \text{Insertion Loss in dB from port 1 to 2}.$
Using the concept of thermodynamic equilibrium:
$$N_o = k\,T\,B = G(N_i + N_{added}) = \frac{1}{2L}(k\,T\,B + N_{added})$$ $$\Rightarrow\; N_{added} = k\,T\,B\,(2L-1)$$ $$\Rightarrow\; T_e = \frac{N_{added}}{k\,B} = (2L-1)\,T$$ $$\Rightarrow\; F = 1 + \frac{N_{added}}{N_i(T_0)} = 1 + \frac{k\,T\,B\,(2L-1)}{k\,T_0\,B} = 1 + (2L-1)\frac{T}{T_0}$$Problem: A Wilkinson divider has quarter-wave arms with 0.5 dB loss each, at $T = 290$ K. Find the noise figure from port 1 to port 2.
$L = 10^{0.05} = 1.122$
$F = 2L = 2\times1.122 = 2.244 = 3.51$ dB
Equivalently: $F\text{(dB)} = 3 + 0.5 = 3.5$ dB (the 3 dB is from the inherent power split).
$P_t$: transmit power, $P_r$: received power
$G_t$: transmit antenna gain, $G_r$: receive antenna gain
$R$: distance between the transmit and receive antenna
The power density radiated by an isotropic transmit antenna ($D = 1 = 0$ dB) at a distance $R$ is given by:
$$S_{avg} = \frac{P_t}{4\pi R^2} \;\;(\text{W/m}^2)$$The power density radiated by an arbitrary transmit antenna with an antenna gain of $G_t$ at a distance $R$ is:
$$S_{avg} = \frac{G_t P_t}{4\pi R^2} = \frac{EIRP}{4\pi R^2} \;\;(\text{W/m}^2)$$where $EIRP = G_t P_t$, representing "effective isotropic radiated power".
The power received by a receive antenna with an effective aperture area is:
$$P_r = S_{avg}\cdot A_e = \frac{G_t P_t}{4\pi R^2}\,A_e \;\;(\text{W})$$For a receive antenna gain:
$$G_r = \frac{4\pi}{\lambda^2}\,A_e$$The Friis equation can be derived from the above two expressions as:
$$P_r = \frac{G_t\,G_r\,\lambda^2}{(4\pi R)^2}\,P_t \;\;(\text{W}) \;\;\Rightarrow\;\; P_r \propto f^{-2},\;\; P_r \propto R^{-2}$$The antenna noise temperature $T_a$ is the equivalent temperature of a matched resistor that would deliver the same noise power $N_i = kT_aB$ to the receiver as the real antenna does. It is not the physical temperature of the antenna metal — it is a weighted blend of two distinct noise sources.(中文:天線雜訊溫度 $T_a$ 是一個 the physical temperature of the antenna metal — it is a weighted blend of two distinct noise sources. (In other words: the antenna noise temperature $T_a$ is an アンテナ金属の物理温度そのものではなく、2 つの異なる雑音源の重み付き混合です。(言い換えれば、アンテナ雑音温度 $T_a$ は等效equivalent等価溫度,代表一個匹配電阻在此溫度下會輸出與實際天線相同的雜訊功率 $N_i = kT_aB$ 給接收機。它 temperature — the temperature at which a matched resistor would deliver the same noise power $N_i = kT_aB$ to the receiver as the real antenna does. It is 温度であり、整合抵抗がその温度にあるとき、実際のアンテナと同じ雑音電力 $N_i = kT_aB$ を受信機へ供給します。これは不是not決して天線金屬的物理溫度,而是兩個不同雜訊來源的加權混合:一是天線指向方向的背景亮度溫度 $T_b$(透過效率 $\eta_{rad}$ 進入),二是天線本體因損耗而自發輻射的物理溫度 $T_p$(以 $1-\eta_{rad}$ 的比例貢獻)。) the physical temperature of the antenna metal, but a weighted blend of two different noise sources: first, the background brightness temperature $T_b$ in the direction the antenna is pointing (entering through the efficiency $\eta_{rad}$), and second, the physical temperature $T_p$ of the antenna body itself, which radiates spontaneously because of its losses (contributing in proportion to $1-\eta_{rad}$).)アンテナ金属の物理温度ではなく、2 つの異なる雑音源の重み付き混合です。第一にアンテナが向いている方向の背景輝度温度 $T_b$(放射効率 $\eta_{rad}$ を通して入ります)、第二にアンテナ本体が損失のために自発的に放射する物理温度 $T_p$($1-\eta_{rad}$ の割合で寄与します)。)
Symbol meanings:
Physical reasoning (reciprocity): A lossy antenna acts like a two-port mixer of noise sources. By reciprocity, only the fraction $\eta_{rad}$ of the incoming sky brightness reaches the terminals — the rest is absorbed by ohmic loss. Simultaneously, the lossy material itself emits thermal noise at its physical temperature $T_p$, and the fraction it contributes to the terminals is exactly $(1-\eta_{rad})$ (the same factor by which it absorbs). Adding these two independent contributions:
$$\underbrace{\eta_{rad}\,T_b}_{\text{sky seen through losses}} \;+\; \underbrace{(1-\eta_{rad})\,T_p}_{\text{self-emission of lossy parts}} \;=\; T_a$$Limiting cases:
Why it matters for receivers: In Friis-equation analysis the input noise is $N_i = kT_aB$, and $T_a$ is plugged directly into the system noise budget $T_{SYS} = T_a + T_e$. For satellite/radio-astronomy links pointed at cold sky, minimizing $(1-\eta_{rad})T_p$ — by using low-loss feed lines, cooled feeds, or higher $\eta_{rad}$ — is often more impactful than reducing the receiver $T_e$ itself.
Common pitfalls:
Reference (rigorous definition of $T_b$): the brightness temperature seen by the antenna is the directivity-weighted average of the surrounding sky brightness $T_B(\theta,\phi)$ over the upper hemisphere:
$T_b = \dfrac{\displaystyle\int_{\phi=0}^{2\pi}\!\!\int_{\theta=0}^{\pi/2} T_B(\theta,\phi)\,D(\theta,\phi)\,\sin\theta\,d\theta\,d\phi}{\displaystyle\int_{\phi=0}^{2\pi}\!\!\int_{\theta=0}^{\pi/2} D(\theta,\phi)\,\sin\theta\,d\theta\,d\phi}$
where $T_B(\theta,\phi)$ is the angular distribution of the background brightness temperature and $D(\theta,\phi)$ is the antenna's directivity pattern. The simple scalar $T_b$ used in $T_a = \eta_{rad}T_b + (1-\eta_{rad})T_p$ is exactly this beam-weighted integral.
Problem: A satellite link at 4 GHz has $P_t = 10$ W, $G_t = 30$ dB, $G_r = 40$ dB, $R = 36{,}000$ km. The receive antenna has $\eta_{rad} = 0.95$, $T_b = 30$ K, $T_p = 290$ K. Find (a) EIRP, (b) received power, (c) antenna noise temperature.
(a) EIRP $= G_t P_t = 10^3 \times 10 = 10{,}000$ W $= 40$ dBW $= 70$ dBm
(b) $\lambda = 3\times10^8/(4\times10^9) = 0.075$ m
$P_r = 10^3\times10^4\times10\times\left(\frac{0.075}{4\pi\times3.6\times10^7}\right)^2$
$= 10^8 \times \left(\frac{0.075}{4.524\times10^8}\right)^2 = 10^8 \times (1.658\times10^{-10})^2 = 10^8\times2.75\times10^{-20}$
$= 2.75\times10^{-12}$ W $= -85.6$ dBm
(c) $T_a = 0.95\times30 + 0.05\times290 = 28.5 + 14.5 = 43$ K
G/T ratio is a useful figure of merit for a receive antenna to evaluate the resulting SNR at the input to a receiver.
where $G_r$ is the gain of the antenna, and $T_a$ is the antenna noise temperature.
To see why $G_r/T_a$ governs the input SNR, combine the Friis transmission equation with the input noise:
For a given link (fixed EIRP, frequency, distance, bandwidth), every other factor is constant — only $G_r/T_a$ depends on the receive antenna. A higher G/T means better sensitivity. When the receiver noise is included:
dBi = decibels relative to an isotropic radiator. An isotropic antenna is a hypothetical reference that radiates equally in all directions (linear gain $= 1$, i.e. $0$ dBi). Real antennas concentrate energy in some direction, so their gain expressed in dBi tells you how much stronger their peak radiation is compared to isotropic.
$$G(\text{dBi}) = 10\log_{10}\!\left(\frac{G_{\text{antenna}}}{G_{\text{isotropic}}}\right) = 10\log_{10}(G)$$Typical values:
vs. dBd: Some antennas (e.g. broadcast) are quoted in dBd, referenced to a half-wave dipole. Conversion: $G(\text{dBi}) = G(\text{dBd}) + 2.15$.
Why dBi here? The Friis equation uses the absolute gain $G_r$ over an isotropic reference, so $G_r(\text{dBi})$ plugs directly into the link budget. In the G/T formula above, since $T$ is on a log scale ($10\log T$ in dBK), $G_r$ must also be on a log scale — and dBi is the natural choice.
Problem. A DBS system operates at 12.2–12.7 GHz (use $f = 12.45$ GHz, $\lambda = 3\times10^8/12.45\times10^9 = 0.0241$ m). It has:
Find: (a) EIRP of transmitter; (b) $G/T$ of the receive antenna w/o and w/ the LNB; (c) CNR at the input and output of the LNB. Then explain why $\text{CNR}_{in} - \text{CNR}_{out} \neq F$.
Linear conversions: $34$ dB $= 2512$, $33.5$ dB $= 2239$, $1.1$ dB $= 1.29$.
(a) EIRP. By definition $\text{EIRP} = G_t P_t$:
$$\text{EIRP} = 2512 \times 120 = 3.01\times10^{5}\text{ W} = 54.8\text{ dBW} \;(=84.8\text{ dBm})$$(b) G/T of receive antenna. For a very-high-gain dish pointed at the cold sky, $T_a \approx T_b$ (sidelobe ground-pickup is negligible).
Without LNB (only antenna noise):
$$G/T \;=\; 10\log\frac{G_r}{T_a} \;=\; 10\log\frac{2239}{50} \;=\; 16.5\text{ dB/K}$$With LNB. Convert NF to noise temperature using Eq. (2): $T_e = (F-1)T_0$:
$$T_e = (1.29 - 1)\times 290 = 0.29\times 290 \approx 84\text{ K}$$The system noise temperature seen at the LNB input is $T_a + T_e = 50 + 84 = 134$ K, so:
$$G/T \;=\; 10\log\frac{G_r}{T_a + T_e} \;=\; 10\log\frac{2239}{134} \;=\; 12.2\text{ dB/K}$$Adding the LNB drops $G/T$ by $\Delta = 16.5 - 12.2 = 4.3$ dB — this is the receiver's contribution to the figure of merit.
(c) CNR at LNB input and output. Use the link equation (Friis transmission combined with $N = kT_{sys}B$):
$$\text{CNR} \;=\; \frac{S_i}{N_i} \;=\; \frac{G_r}{T_{sys}} \cdot \frac{\text{EIRP}\;\lambda^2}{kB\,(4\pi R)^2}$$Input of LNB ($T_{sys} = T_a$, no receiver noise yet):
$$\text{CNR}_{in} = \frac{2239}{50}\cdot\frac{3.01\times10^5\,(0.0241)^2}{1.38\times10^{-23}\times 20\times10^6\times(4\pi\times3.9\times10^7)^2} = 118.1 = 20.7\text{ dB}$$Output of LNB — gain cancels, but the noise floor is now $kB(T_a+T_e)$:
$$\text{CNR}_{out} = \frac{G_{LNB}\,S_i}{G_{LNB}\,kB(T_a+T_e)} = \frac{G_r}{T_a+T_e}\cdot\frac{\text{EIRP}\;\lambda^2}{kB(4\pi R)^2} = 44.1 = 16.4\text{ dB}$$Why is $\text{CNR}_{in} - \text{CNR}_{out} = 4.3$ dB, not $F = 1.1$ dB?
The textbook identity "SNR drop = noise figure" assumes $N_i = kT_0B$ (i.e. source at the standard 290 K) — that is the very definition of $F$. Here the source noise is the cold sky $T_a = 50$ K, not 290 K, so $F$ no longer equals the SNR-degradation:
$$\text{CNR drop} = 10\log\frac{T_a + T_e}{T_a} = 10\log\frac{134}{50} = 4.3\text{ dB}$$ $$\text{vs. } F\text{ (defined at } T_0 = 290\text{ K)} = 10\log\frac{T_0 + T_e}{T_0} = 10\log\frac{374}{290} = 1.1\text{ dB}$$When $T_a < T_0$, the receiver's $T_e$ looks proportionally larger compared to the cold input noise, so the SNR penalty is bigger than $F$ would suggest. This is exactly why satellite/radio-astronomy receivers use $G/T$ and noise temperature, not $F$, as the figure of merit.
For a worked example where $T_a \neq T_0$ is handled correctly through a full receiver cascade (LNA → BPF → Mixer), see Example — Analysis of Wireless Receiver. It uses $T_a = 150$ K (not 290 K) and computes the output noise via $N_o = G\,kB(T_a + T_e)$ instead of the (incorrect-here) $N_o = G\,kT_0BF$.
LNB = Low-Noise Block downconverter — the small device mounted at the focal point of a satellite dish (the part the coax cable plugs into). It bundles three RF stages in one weather-sealed package:
| Stage | Purpose |
|---|---|
| LNA (low-noise amp) | Boost the very weak microwave signal ($\sim -100$ dBm) before any cable loss adds noise — this is what sets the system noise figure (Friis: first stage dominates). |
| BPF (band-pass filter) | Reject out-of-band signals and image frequencies before mixing. |
| Mixer + LO | Down-convert the satellite RF (Ku-band 12.2–12.7 GHz) to a lower IF (typically 950–2150 MHz) so it can travel down cheap coax to the indoor receiver without huge loss. |
Why "Block"? It down-converts an entire block of frequencies (the whole 500 MHz Ku slice) at once, not channel-by-channel — channel selection happens later in the indoor set-top box.
Why mounted at the dish? Two reasons, both about noise:
In this example: the LNB has $F = 1.1$ dB (very low — typical for a modern Ku-band LNB, giving $T_e \approx 84$ K). Because it is the first amplifier and has high gain, by Friis it is the dominant noise contributor — everything after it is suppressed by $G_{LNB}$.
CNR is the ratio of the received carrier (signal) power to the noise power at the receiver input (or demodulator input). It is the link-budget version of "SNR" used at RF/IF — i.e. before demodulation, while the signal is still a modulated carrier.
$$\text{CNR} \;=\; \frac{C}{N} \;=\; \frac{S_i}{k\,T_{sys}\,B}$$In dB form, with all link components added:
$$\text{CNR (dB)} \;=\; \underbrace{\text{EIRP}}_{\text{transmitter}} \;+\; \underbrace{G/T}_{\text{receiver}} \;-\; \underbrace{\text{FSPL}}_{\text{free-space loss}} \;-\; 10\log_{10}(kB)$$Symbol meanings:
CNR vs. SNR vs. $E_b/N_0$:
Why we care: CNR sets the demodulator's BER. For a fixed link (EIRP, frequency, distance, bandwidth), CNR depends only on the receive antenna's $G/T$ — which is exactly why G/T is the antenna figure of merit. A 3 dB increase in G/T directly raises CNR by 3 dB, allowing higher-order modulation, lower BER, or smaller transmit power.
Problem: A ground station has $G_r = 45$ dBi, $T_a = 25$ K, and a receiver with $T_e = 50$ K. Find G/T. If EIRP $= 60$ dBW, $f = 12$ GHz, $R = 40{,}000$ km, $B = 36$ MHz, find the CNR.
$G/T = 45 - 10\log(25+50) = 45 - 18.75 = 26.25$ dB/K
$\lambda = 0.025$ m. Free-space loss $= 20\log(4\pi R/\lambda) = 20\log(4\pi\times4\times10^7/0.025) = 20\log(2.01\times10^{10}) = 206.1$ dB
$\text{CNR} = \text{EIRP} + G/T - \text{FSPL} - 10\log(kB)$
$10\log(kB) = 10\log(1.38\times10^{-23}\times36\times10^6) = 10\log(4.97\times10^{-16}) = -153.0$ dBW
$\text{CNR} = 60 + 26.25 - 206.1 - (-153.0) = 60 + 26.25 - 206.1 + 153.0 = 33.15$ dB
Correct approach: $N_{added} = (F-1)kT_0B$, then $N_o = G[kT_aB + N_{added}] = GkB[T_a + T_e]$
Problem: In the previous example for an IS-54 cellular system using QPSK modulation, the receive antenna has a noise temperature of 900 K and the receiver has a noise figure of 8 dB. Find the minimum detectable signal level (sensitivity).
Given $(S_o/N_o)_{min} = 11.7$ dB $= 14.8$, $\;\;F = 8$ dB $= 6.3$
Step 1 — Identify the key formula. The minimum detectable signal requires that the output SNR meets the modulation threshold (the bracket $T_a + (F-1)T_0 = T_a + T_e$ uses Eq. (2)):
$$S_{i,min} = kB\bigl[T_a + (F-1)T_0\bigr]\left(\frac{S_o}{N_o}\right)_{min}$$Step 2 — Convert noise figure to linear.
$F = 8$ dB $= 10^{0.8} = 6.31$, so $F - 1 = 5.31$
Step 3 — Compute the effective noise temperature seen at the input.
$T_a + (F-1)T_0 = 900 + 5.31\times 290 = 900 + 1540 = 2440$ K
Step 4 — Noise power over the bandwidth.
$kB \cdot 2440 = 1.38\times10^{-23}\times 30\times10^3\times 2440 = 1.010\times10^{-15}$ W
Step 5 — Required output SNR.
$(S_o/N_o)_{min} = 11.7$ dB $= 10^{1.17} = 14.8$
Step 6 — Minimum detectable signal.
$S_{i,min} = 1.010\times10^{-15}\times 14.8 = 1.49\times10^{-14}$ W $= \boxed{-108.3\text{ dBm}}$
The bracket is the system noise temperature at the receiver input, $T_{sys} = T_a + (F-1)T_0$. Each number maps to one physical piece:
The product $(F-1)T_0 = 5.31 \times 290 \approx 1540$ K is the equivalent noise temperature of the receiver, $T_e$. This conversion exists because $F$ is defined at $T_0 = 290$ K, but to add receiver noise to antenna noise on equal footing you need it expressed as a temperature:
$$T_e = (F-1)T_0$$Both $T_a$ and $T_e$ appear at the same input node, so they simply add:
$$T_{sys} = T_a + T_e = 900 + 1540 = 2440\text{ K}$$That total is what goes into $N = kB\,T_{sys}$ for the input-referred noise power in Step 4.
Setup. A QPSK symbol carries 2 bits. Split them into an in-phase bit $b_I\in\{\pm 1\}$ and a quadrature bit $b_Q\in\{\pm 1\}$. The transmitted waveform is
$$s(t)=b_I\cos(\omega_c t)-b_Q\sin(\omega_c t)$$That single line is already the answer — it is a sum of two BPSK signals, one riding $\cos$, the other riding $\sin$.
Why the two streams don't interfere — orthogonality. Over one symbol period $T_s$,
$$\int_0^{T_s}\cos(\omega_c t)\sin(\omega_c t)\,dt = 0$$so $\cos$ and $\sin$ are orthogonal basis functions. At the receiver:
The two bits decouple cleanly; each branch is a textbook BPSK detector.
Geometric view (constellation). Treat $\cos$ and $\sin$ as orthogonal axes. The 4 QPSK symbols sit at $(\pm 1,\pm 1)$ — two BPSK constellations on perpendicular axes. Nearest-neighbor distance along each axis equals the BPSK distance, so the per-axis decision is identical to BPSK.
Consequences that follow for free:
QPSK doubles spectral efficiency for free — the I and Q channels share the same RF bandwidth with no crosstalk, so orthogonality buys you the second BPSK channel inside the same spectrum.
Caveat — coherent demodulation required. The receiver must lock to the carrier phase so its $\cos$ and $\sin$ match the transmitter's. A 90° phase error swaps $I$ and $Q$; a 45° error mixes them and destroys the orthogonality. That is why QPSK receivers need a Costas loop (or pilot / preamble) for carrier recovery.
Smallest input signal $S_{i,min}$ that still meets the demodulator's threshold $(S_o/N_o)_{min}$.
Build-up from $\text{SNR}_{out} = G\,S_i / N_o$:
What each factor controls:
| Term | Meaning | How to reduce it |
|---|---|---|
| $kB$ | Thermal-noise floor over bandwidth $B$ | narrower $B$ (matched filter) |
| $T_a$ | Antenna / sky noise | point at cold sky, low side-lobes |
| $(F-1)T_0$ | Receiver-added noise | low-NF LNA first (Friis) |
| $(S_o/N_o)_{min}$ | Modulation requirement | better coding, lower-order modulation |
Intuition: sensitivity = (total input-referred noise power) × (required SNR margin). Halve the noise temperature → 3 dB better sensitivity. Halve the bandwidth → 3 dB better. Drop the required SNR by 3 dB (better FEC) → 3 dB better.
The bandwidth $B = 30\times10^3$ Hz in Step 4 is not introduced in this example — it is inherited from the prior IS-54 QPSK lecture example, where $R_b = 48.6$ kbps and $B = 30$ kHz are stated up-front. The current problem opens with "In the previous example for an IS-54 cellular system using QPSK modulation...", so $B$ carries over.
Why specifically 30 kHz? IS-54 (a.k.a. D-AMPS / TDMA-54) is the early-1990s North-American digital cellular standard. It re-uses the analog AMPS 30 kHz channel grid mandated by the FCC — each RF carrier occupies a 30 kHz slot. Three users share that carrier via TDMA, but the per-carrier RF bandwidth stays fixed at 30 kHz.
Putting numbers into $kB\,T_{sys}$:
For other standards just swap $B$: GSM uses 200 kHz, LTE uses 1.4–20 MHz, Wi-Fi 802.11 uses 20/40/80 MHz — and the noise floor $kBT_{sys}$ scales linearly with whichever channel bandwidth the radio actually filters.
Problem: In a simplified wireless receiver, the input noise power from a feeding antenna is $N_i = k\,T_a\,B$, where $T_a = 150$ K. Assume the system is at temperature $T_0 = 290$ K, with a characteristic impedance of 50 $\Omega$, and an IF bandwidth of $B = 10$ MHz.
→ For the full derivation of $F = L$ on a lossy line, see §2.2.4 Noise Figure of a Lossy Line.
$L$ = Insertion loss (in dB) — the power lost as the signal passes through a passive stage:
$$L_{\text{dB}} = 10\log_{10}\!\left(\frac{P_{\text{in}}}{P_{\text{out}}}\right) \;\;\Longleftrightarrow\;\; G = \frac{1}{L} = 10^{-L_{\text{dB}}/10}$$For passive components (filters, attenuators, cables, passive mixers), gain $G < 1$, so it's more natural to quote the loss $L > 1$ instead. The two are reciprocals.
In this diagram:
| Stage | Loss / Gain |
|---|---|
| BPF: $L = 1$ dB | $G_2 = 10^{-0.1} = 0.79$ — loses 21% of signal power |
| Mixer: $L = 3$ dB | $G_3 = 10^{-0.3} = 0.50$ — loses 50% (typical of a passive diode mixer) |
Why both $L$ and $F$ for the mixer? $L$ tells you how much signal is lost; $F$ tells you how much extra noise is added on top of the thermal floor. They are independent specs — a mixer can have low loss but high noise figure if it has internal noise sources (LO leakage, port mismatch, etc.).
Key relation for passive lossy stages at room temperature ($T = T_0$):
$$F = L \quad\text{(in dB: NF = insertion loss)}$$That's why for the BPF only $L$ is given — its noise figure is implicitly $F = L = 1$ dB. The mixer breaks this rule because it's an active/non-equilibrium component (LO drives it), so its $F = 4$ dB exceeds its $L = 3$ dB.
For the Friis cascade above, each $L$ is converted to gain: $G_2 = 1/L_2 = 0.79$, $G_3 = 1/L_3 = 0.50$ — that's where the linear values in part (a) come from.
(a) Find the cascaded parameters $F$, $T_e$, and $G_{total}$.
(b) Find the output noise power and two-sided power spectral density.
(c) Find the minimum detectable power and voltage for the input signal under a minimum detectable SNR of 20 dB at the output of the receiver.
(a) Cascaded parameters $F$, $T_e$, $G_{total}$
Convert each stage to linear:
LNA: $F_1 = 2$ dB $= 1.58$, $G_1 = 10$ dB $= 10$
BPF: $F_2 = L = 1$ dB $= 1.26$, $G_2 = -1$ dB $= 0.79$
Mixer: $F_3 = 4$ dB $= 2.51$, $G_3 = -3$ dB $= 0.5$
Cascade noise figure (Friis formula):
$$F = F_1 + \frac{F_2 - 1}{G_1} + \frac{F_3 - 1}{G_1 G_2} = 1.58 + \frac{0.26}{10} + \frac{1.51}{10\times 0.79} = 1.58 + 0.026 + 0.191 = \boxed{1.80\;(2.55\text{ dB})}$$From Eq. (1): $T_e = (F-1)\,T_0 = 0.80\times 290 = \boxed{232\text{ K}}$
$G_{total} = G_1 \cdot G_2 \cdot G_3 = 10\times 0.79\times 0.5 = \boxed{3.95\;(5.97\text{ dB})}$
(b) Output noise power and two-sided PSD
Input noise: $N_i = k\,T_a\,B = 1.38\times10^{-23}\times 150\times 10^7 = 2.07\times10^{-14}$ W (where $B = 10\text{ MHz} = 10\times 10^6 = 10^7$ Hz, from the problem statement)
Output noise includes amplified input noise plus receiver-added noise:
$$N_o = G_{total}\,k\,B\,(T_a + T_e) = 3.95\times 1.38\times10^{-23}\times 10^7\times(150+232)$$$= 3.95\times 1.38\times10^{-16}\times 382 = \boxed{2.08\times10^{-13}\text{ W}\;(-96.8\text{ dBm})}$
Two-sided PSD (noise is symmetric about $f = 0$, so divide by $2B$):
$$S_{no} = \frac{N_o}{2B} = \frac{2.08\times10^{-13}}{2\times10^7} = \boxed{1.04\times10^{-20}\text{ W/Hz}}$$(c) Minimum detectable signal for $\text{SNR}_{out} = 20$ dB
$\text{SNR}_{out,min} = 20$ dB $= 100$ (linear)
At the output: $\text{SNR}_{out} = G_{total}\,S_{i,min} / N_o$, so:
$$S_{i,min} = \frac{\text{SNR}_{out,min}\cdot N_o}{G_{total}} = \frac{100\times 2.08\times10^{-13}}{3.95} = \boxed{5.27\times10^{-12}\text{ W}\;(-82.8\text{ dBm})}$$Minimum detectable voltage (across $Z_0 = 50\;\Omega$):
$$V_{i,min} = \sqrt{S_{i,min}\cdot Z_0} = \sqrt{5.27\times10^{-12}\times 50} = \sqrt{2.63\times10^{-10}} = \boxed{16.2\;\mu\text{V}}$$Problem: For the following wireless receiver, if the received power at the antenna terminal is $S_i = -80$ dBm, calculate the input and output SNR. This is the worked example from the Chap 2 slides; the condensed slide derivation and the full step-by-step solution are merged below.
The receiver chain includes: Background noise ($T_b$) → Antenna ($G_A$, $\eta_{rad}$, $T_P$) → Transmission line ($L_T$, $T_P$) → RF Amp. ($G_{RF}$, $T_{RF}$) → Mixer with LO ($L_M$, $T_M$) → IF Amp. ($G_{IF}$, $T_{IF}$) → Output ($N_o$, $S_o$). The input signal and noise at the antenna terminal are $N_i$, $S_i$.
Given: $f = 4$ GHz, $B = 1$ MHz, $\eta_{rad}=0.9$, $T_p=300$ K, $T_b=200$ K, $L_T=1.5$ dB, $G_{RF}=20$ dB, $F_{RF}=3$ dB, $L_M=6$ dB, $F_M=7$ dB, $G_{IF}=30$ dB, $F_{IF}=1.1$ dB.
Step 1 — Antenna noise temperature.
$T_A = \eta_{rad}\,T_b + (1-\eta_{rad})\,T_P = 0.9\times 200 + 0.1\times 300 = 180 + 30 = 210$ K
Step 2 — Transmission line noise contribution.
$L_T = 1.5$ dB $= 10^{0.15} = 1.413$
$T_{TL} = (L_T - 1)\,T_P = (1.413 - 1)\times 300 = 0.413\times 300 = 123$ K
Step 3 — Receiver cascade noise temperature.
Convert component values to linear:
$G_{RF} = 10^{2.0} = 100$, $\;F_{RF} = 10^{0.3} = 2.0$, $\;L_M = 10^{0.6} = 4$, $\;F_M = 10^{0.7} = 5.01$, $\;G_{IF} = 10^{3.0} = 1000$, $\;F_{IF} = 10^{0.11} = 1.29$
Individual noise temperatures:
$T_{RF} = (F_{RF}-1)\,T_0 = (2.0-1)\times 290 = 289$ K
$T_M = (F_M - 1)\,T_0 = (5.01-1)\times 290 = 1163$ K (where $F_M = 7$ dB $\Rightarrow 10^{7/10} = 10^{0.7} = 5.01$ linear, from the given values)
$T_{IF} = (F_{IF}-1)\,T_0 = (1.29-1)\times 290 = 84$ K
Friis cascade (RF amp → Mixer → IF amp):
$\color{#b44a2f}{T_{REC} = T_{RF} + \dfrac{T_M + T_{IF}\cdot L_M}{G_{RF}} = 289 + \dfrac{1163 + 84\times 4}{100} = 289 + \dfrac{1499}{100} = 289 + 15 = 304}$ K
The general Friis cascade for noise temperature (see eq.) is:
$$T_e = T_{e1} + \dfrac{T_{e2}}{G_1} + \dfrac{T_{e3}}{G_1\,G_2} + \cdots$$Map the three stages of this receiver onto the formula:
| Stage | Role | Gain $G_i$ | Noise temp $T_{ei}$ |
|---|---|---|---|
| 1 | RF amp | $G_1 = G_{RF}$ | $T_{e1} = T_{RF}$ |
| 2 | Mixer (passive lossy) | $G_2 = 1/L_M$ | $T_{e2} = T_M$ |
| 3 | IF amp | $G_3 = G_{IF}$ | $T_{e3} = T_{IF}$ |
Substitute:
$$T_{REC} = T_{RF} + \dfrac{T_M}{G_{RF}} + \dfrac{T_{IF}}{G_{RF}\cdot(1/L_M)} = T_{RF} + \dfrac{T_M}{G_{RF}} + \dfrac{T_{IF}\cdot L_M}{G_{RF}}$$The mixer's loss $L_M$ pops up to the numerator because dividing by $G_2 = 1/L_M$ is the same as multiplying by $L_M$. Combine the last two terms over the common denominator $G_{RF}$:
$$T_{REC} = T_{RF} + \dfrac{T_M + T_{IF}\cdot L_M}{G_{RF}}$$Physical meaning: the IF amp's noise is referred back to the input through the RF amp's gain and through the mixer's loss — a passive lossy stage hurts everything downstream because it attenuates the signal but the noise still has to be referred back through it.
Step 4 — System noise temperature.
$T_{SYS} = T_A + T_{TL} + L_T\cdot T_{REC} = 210 + 123 + 1.413\times 304 = 210 + 123 + 429 = 762$ K
$T_{SYS}$ is being expressed referred to the antenna terminal — i.e., as if all noise sources were located at the antenna input. The transmission line sits between the antenna and the receiver, with gain $G_T = 1/L_T$ (loss $L_T \ge 1$).
Apply the Friis cascade for noise temperature with the line as stage 1 and the receiver as stage 2:
$$T_{e,\text{cascade}} = T_{TL} + \dfrac{T_{REC}}{G_T} = T_{TL} + \dfrac{T_{REC}}{1/L_T} = T_{TL} + L_T\cdot T_{REC}$$The antenna noise $T_A$ is already at the input, so it just adds:
$$\boxed{\,T_{SYS} = T_A + T_{TL} + L_T\cdot T_{REC}\,}$$Intuition: the line attenuates the signal by $1/L_T$ on its way to the receiver. So when we push the receiver's noise back through the line (toward the antenna), it has to be scaled up by $L_T$ to keep the same SNR — a small amount of noise generated past a 3 dB attenuator looks like twice that much noise at the attenuator's input.
Numerical check: $L_T = 1.5\text{ dB} = 1.413$, $T_{REC} = 304$ K $\Rightarrow L_T \cdot T_{REC} = 429$ K. The line itself contributes only 123 K, but it inflates the receiver's effective contribution from 304 K to 429 K — a 41% penalty just for sitting in front of the receiver.
Step 5 — Input SNR.
$N_i = k\,B\,T_A = 1.38\times10^{-23}\times 10^6\times 210 = 2.90\times10^{-15}$ W $= -115.4$ dBm
$\text{SNR}_{in} = S_i - N_i = -80 - (-115.4) \approx -80 - (-115) = 35$ dB
Step 6 — Output SNR.
$N_{SYS} = k\,B\,T_{SYS} = 1.38\times10^{-23}\times 10^6\times 762 = 1.05\times10^{-14}$ W $= -109.8$ dBm
$\text{SNR}_{out} = \dfrac{S_i}{k\,B\,T_{SYS}} = -80 - (-109.8) \approx -80 - (-110) = 30$ dB
Result: $\text{SNR}_{in} \approx \boxed{35\text{ dB}}$, $\;\text{SNR}_{out} \approx \boxed{30\text{ dB}}$
The SNR degrades by about 5 dB through the system. The dominant noise contributors are the receiver chain ($L_T \cdot T_{REC} = 429$ K, 56%) and the antenna ($T_A = 210$ K, 28%), with the transmission line adding $T_{TL} = 123$ K (16%).
Problem: Repeat the Minimum Detectable Signal Level example (IS-54 system, $T_a = 900$ K, $F = 8$ dB, $B = 30$ kHz) for (a) BPSK with $R_b = 48.6$ kbps, and (b) 64-QAM with $R_b = 48.6$ kbps. Find the minimum detectable input signal power for each. Assume $\color{#b44a2f}{\text{BER} = 10^{-5}}$ for both.
Pozar Table 14.1 reference values: For BER $= 10^{-5}$: BPSK requires $E_b/n_0 = 9.6$ dB; 64-QAM requires $E_b/n_0 \approx 22.5$ dB. BPSK spectral efficiency $R_b/B = 1$ bps/Hz; 64-QAM spectral efficiency $R_b/B = 6$ bps/Hz.
Common parameters: $F = 8$ dB $= 6.31$, $T_a = 900$ K, $B = 30$ kHz, $R_b = 48.6$ kbps.
$S_{i,min} = kB[T_a + (F-1)T_0] \cdot (S_o/N_o)_{min}$, where $(S_o/N_o)_{min} = (E_b/n_0)\cdot(R_b/B)$.
The bracket $T_a + (F-1)T_0$ is the system noise temperature at the receiver input, $T_{sys}$. Each number maps to one physical piece:
The product $(F-1)T_0 = 5.31 \times 290 \approx 1540$ K is the equivalent noise temperature of the receiver, $T_e$. This conversion exists because $F$ is defined at $T_0 = 290$ K, but to add receiver noise to antenna noise on equal footing it must be expressed as a temperature:
$$T_e = (F-1)T_0$$Both $T_a$ and $T_e$ appear at the same input node, so they simply add:
$$T_{sys} = T_a + T_e = 900 + 1540 = 2440\text{ K}$$That total is what goes into $N = kB\,T_{sys}$ for the input-referred noise power — the bracket $[900 + 5.31\times 290]$ in the BPSK and 64-QAM lines below is exactly this $T_{sys}$.
(a) BPSK:
$E_b/n_0 = 9.6$ dB $= 9.12$ (linear). Spectral efficiency $R_b/B = 48.6/30 = 1.62$.
$(S_o/N_o)_{min} = 9.12 \times 1.62 = 14.8$ → $= 11.7$ dB (same as QPSK!)
$S_{i,min} = 1.38\times10^{-23}\times30\times10^3\times[900 + 5.31\times290]\times14.8$
$= 4.14\times10^{-19}\times2440\times14.8 = 1.49\times10^{-14}$ W $= \boxed{-108.3\text{ dBm}}$
Note: BPSK and QPSK have the same $E_b/n_0$ requirement and (for the same $R_b/B$ ratio) the same sensitivity. This makes sense because QPSK is essentially two orthogonal BPSK channels.
(b) 64-QAM:
$E_b/n_0 = 22.5$ dB $= 177.8$ (linear). Spectral efficiency $R_b/B = 48.6/30 = 1.62$.
$(S_o/N_o)_{min} = 177.8 \times 1.62 = 288.0$ → $= 24.6$ dB
$S_{i,min} = 1.38\times10^{-23}\times30\times10^3\times[900 + 5.31\times290]\times288.0$
$= 4.14\times10^{-19}\times2440\times288.0 = 2.91\times10^{-13}$ W $= \boxed{-95.4\text{ dBm}}$
64-QAM requires about 12.9 dB more signal power than BPSK/QPSK to achieve the same BER. This is the cost of higher spectral efficiency — 64-QAM packs 6 bits per symbol, so constellation points are much closer together and more vulnerable to noise.
Summary comparison:
| Modulation | $E_b/n_0$ (dB) | $\eta = R_b/B$ | $S_o/N_o$ (dB) | $S_{i,min}$ (dBm) |
|---|---|---|---|---|
| BPSK | 9.6 | 1.62 | 11.7 | -108.3 |
| QPSK | 9.6 | 1.62 | 11.7 | -108.3 |
| 64-QAM | 22.5 | 1.62 | 24.6 | -95.4 |
Setup. A QPSK symbol carries 2 bits. Split them into an in-phase bit $b_I\in\{\pm 1\}$ and a quadrature bit $b_Q\in\{\pm 1\}$. The transmitted waveform is
$$s(t)=b_I\cos(\omega_c t)-b_Q\sin(\omega_c t)$$That single line is already the answer — it is a sum of two BPSK signals, one riding $\cos$, the other riding $\sin$.
Why the two streams don't interfere — orthogonality. Over one symbol period $T_s$,
$$\int_0^{T_s}\cos(\omega_c t)\sin(\omega_c t)\,dt = 0$$so $\cos$ and $\sin$ are orthogonal basis functions. At the receiver:
The two bits decouple cleanly; each branch is a textbook BPSK detector.
Geometric view (constellation). Treat $\cos$ and $\sin$ as orthogonal axes. The 4 QPSK symbols sit at $(\pm 1,\pm 1)$ — two BPSK constellations on perpendicular axes. Nearest-neighbor distance along each axis equals the BPSK distance, so the per-axis decision is identical to BPSK.
Consequences that follow for free:
QPSK doubles spectral efficiency for free — the I and Q channels share the same RF bandwidth with no crosstalk, so orthogonality buys you the second BPSK channel inside the same spectrum.
Caveat — coherent demodulation required. The receiver must lock to the carrier phase so its $\cos$ and $\sin$ match the transmitter's. A 90° phase error swaps $I$ and $Q$; a 45° error mixes them and destroys the orthogonality. That is why QPSK receivers need a Costas loop (or pilot / preamble) for carrier recovery.
Problem: Use the wireless receiver from the full receiver analysis example (Section 2.9: $f=4$ GHz, $B=1$ MHz, $\eta_{rad}=0.9$, $T_p=300$ K, $T_b=200$ K, $L_T=1.5$ dB, $G_{RF}=20$ dB, $F_{RF}=3$ dB, $L_M=6$ dB, $F_M=7$ dB, $G_{IF}=30$ dB, $F_{IF}=1.1$ dB) in a point-to-point microwave link. The transmitter has EIRP $= 30$ dBW, $R = 30$ km, and the receive antenna gain is $G_r = 25$ dBi. Find (a) the received signal power $S_i$, (b) the system noise temperature $T_{SYS}$, (c) the input and output SNR.
(a) Received signal power via Friis equation:
$\lambda = c/f = 3\times10^8/(4\times10^9) = 0.075$ m
Free-space path loss: $\text{FSPL} = 20\log(4\pi R/\lambda) = 20\log(4\pi\times3\times10^4/0.075) = 20\log(5.03\times10^6) = 134.0$ dB
$S_i = \text{EIRP} + G_r - \text{FSPL} = 30 + 25 - 134.0 = -79.0$ dBW $= -49.0$ dBm
(b) System noise temperature (from the Section 2.9 example, already computed):
$T_A = 0.9\times200 + 0.1\times300 = 210$ K
$T_{TL} = (L_T - 1)T_p = (1.413-1)\times300 = 123$ K
$T_{REC} = T_{RF} + \frac{T_M + T_{IF}\cdot L_M}{G_{RF}} = 289 + \frac{1163 + 84\times4}{100} = 289 + 15 = 304$ K (Friis cascade)
$T_{SYS} = T_A + T_{TL} + L_T\cdot T_{REC} = 210 + 123 + 1.413\times304 = 762$ K
(c) Input SNR:
$N_i = kBT_A = 1.38\times10^{-23}\times10^6\times210 = 2.90\times10^{-15}$ W $= -115.4$ dBm
$\text{SNR}_{in} = S_i/N_i = -49.0 - (-115.4) = 66.4$ dB
Output SNR:
$N_{SYS} = kBT_{SYS} = 1.38\times10^{-23}\times10^6\times762 = 1.05\times10^{-14}$ W $= -109.8$ dBm
$\text{SNR}_{out} = S_i/kBT_{SYS} = -49.0 - (-109.8) = \boxed{60.8\text{ dB}}$
The SNR degradation from input to output is $66.4 - 60.8 = 5.6$ dB, which reflects the total system noise contributions from the transmission line and receiver chain. This is a very healthy link margin — even 64-QAM (requiring ~24.6 dB SNR) would work comfortably.
Short answer: using $G/T$ wouldn't be more correct, only a different bookkeeping for the same equation. In this problem $S_i$ is already given at the receiver input port, so the antenna gain has already been "spent" — folding it back in via $G/T$ would double-count it.
Where $G/T$ comes from. The full link-budget form for the carrier-to-noise ratio is
$$\frac{C}{N_0}\;=\;\frac{\mathrm{EIRP}}{L_{path}}\cdot\frac{G_r}{T_{sys}}\cdot\frac{1}{k}$$The receiver-side ratio $G_r/T_{sys}$ — the figure of merit — is grouped together because both terms belong to the receiving station:
A station with higher $G/T$ collects more signal and adds less noise, captured in one number.
Why this problem doesn't need it. Here $S_i = -49$ dBm is the signal power at the receiver input port, after the antenna. Tracing the chain:
$$S_i=\frac{\mathrm{EIRP}\cdot G_r}{L_{path}}\quad\Longrightarrow\quad \frac{S_i}{N_0}=S_i\cdot\frac{1}{kT_{sys}}$$The $G_r$ has already been absorbed into $S_i$ (see Step (a), where $G_r = 25$ dBi was added to EIRP and FSPL was subtracted to get $S_i$). Multiplying by $G/T$ now would apply the antenna gain twice. So the correct formula given a known $S_i$ is exactly what Step (c) uses:
$$\mathrm{SNR}_{out}=\frac{S_i}{kBT_{sys}}$$When $G/T$ is the right tool:
Sanity check. If you wanted to back out an effective $G/T$ from this problem:
$$G/T\;[\text{dB/K}]=G_r\,[\text{dBi}]-10\log_{10}T_{sys}=25-10\log_{10}(762)=25-28.8=-3.8\ \text{dB/K}$$The 60.8 dB output SNR is unchanged — same physics, two different ways of grouping the same factors. So Step (c) is correct as written; $G/T$ would just be a re-derivation route, not a more accurate one.
$a_1$ = small-signal gain, $a_2$ → second harmonics/DC offset, $a_3$ → gain compression and IM3.
➤For an arbitrary input signal of certain bandwidth, the resulting in-band intermodulation products from a similar mechanism will cause distortion of the output signal. This effect is called third-order intermodulation distortion.
Assume that the input signal is a single frequency sinusoid (one tone):
Effective voltage gain at $\omega_0$:
The two identities used throughout this derivation:
$$\cos^2 x = \tfrac{1}{2}\bigl(1 + \cos 2x\bigr) \qquad\text{and}\qquad \cos^3 x = \tfrac{3}{4}\cos x + \tfrac{1}{4}\cos 3x$$These convert powers of a cosine into a sum of cosines at integer multiples of the frequency — the mechanism by which an $n$th-order nonlinearity produces the $n$th harmonic plus a back-contribution to the fundamental (gain compression).
Substitute $v_i = V_0\cos\omega_0 t$ into the Taylor series and apply the identities above:
$$\begin{aligned} v_o(t) &= a_0 + a_1 V_0\cos\omega_0 t + a_2 V_0^2\cos^2\omega_0 t + a_3 V_0^3\cos^3\omega_0 t + \cdots \\[4pt] &= \underbrace{\left(a_0+\tfrac{1}{2}a_2 V_0^2\right)}_{\text{DC}} + \underbrace{\left(a_1 V_0+\tfrac{3}{4}a_3 V_0^3\right)\cos\omega_0 t}_{\text{at }\omega_0} \\ &\quad + \underbrace{\tfrac{1}{2}a_2 V_0^2\cos 2\omega_0 t}_{\text{2nd harmonic}} + \underbrace{\tfrac{1}{4}a_3 V_0^3\cos 3\omega_0 t}_{\text{3rd harmonic}} + \cdots \end{aligned}$$Keeping only Taylor terms up to third order, the component at the input frequency $\omega_0$ has amplitude $a_1 V_0 + \tfrac{3}{4}a_3 V_0^3$. The voltage gain at $\omega_0$ is the ratio of that output amplitude to $V_0$:
$$G_v \;=\; \frac{v_o^{\omega_0}}{v_i^{\omega_0}} \;=\; \frac{a_1 V_0 + \tfrac{3}{4}a_3 V_0^3}{V_0} \;=\; a_1 + \tfrac{3}{4}a_3 V_0^2$$Two observations follow directly:
Since $a_3 < 0$ (typically), gain decreases with increasing $V_0$ → gain compression.
The 1 dB compression point $P_{1\text{dB}}$ is where actual output is 1 dB below the ideal linear extension.
Problem: An amplifier has a small-signal gain of 15 dB. The datasheet specifies $OP_{1\text{dB}} = 22$ dBm. Find (a) $IP_{1\text{dB}}$, and (b) the actual output power at the 1 dB compression point compared to an ideal amplifier.
(a) $IP_{1\text{dB}} = OP_{1\text{dB}} - G = 22 - 15 = 7$ dBm
In the linear region of the amplifier, gain in dB simply adds to input power (in dBm):
which is just the dB-form of the linear power law $P_{\text{out}} = G\cdot P_{\text{in}}$. Solving for the input:
$$IP_{1\text{dB}} = OP_{1\text{dB}} - G = 22 - 15 = 7\ \text{dBm}$$Why no $+1$ dB correction here? Strictly, at the compression point the actual gain is $G - 1$ dB (that's the definition of 1 dB compression), so the exact relation is
$$OP_{1\text{dB}} = IP_{1\text{dB}} + (G - 1) \;\Longrightarrow\; IP_{1\text{dB}} = OP_{1\text{dB}} - G + 1 = 8\ \text{dBm}.$$This problem uses the common simplified form $IP_{1\text{dB}} = OP_{1\text{dB}} - G$ (ignoring the 1 dB compression of the gain itself). Both conventions appear in textbooks — the 1 dB discrepancy is usually small compared to other approximations. See part (b) for the convention-aware version.
中文:在線性區,輸出功率 (dBm) = 輸入功率 (dBm) + 增益 (dB)。在 1 dB 壓縮點將同一線性關係延用,便得 $IP_{1\text{dB}} = OP_{1\text{dB}} - G$。若嚴格考慮壓縮處實際增益為 $G - 1$ dB,正確結果應為 $OP_{1\text{dB}} - G + 1 = 8$ dBm;此處採用簡化慣例不加 1。Note: in the linear region, output power (dBm) = input power (dBm) + gain (dB). Carrying that same linear relation over to the 1 dB compression point gives $IP_{1\text{dB}} = OP_{1\text{dB}} - G$. If one accounts strictly for the fact that the actual gain at compression is $G - 1$ dB, the correct result would be $OP_{1\text{dB}} - G + 1 = 8$ dBm; here we follow the simplified convention and do not add the 1.注:線形領域では 出力電力 (dBm) = 入力電力 (dBm) + 利得 (dB) です。同じ線形関係を 1 dB 圧縮点にそのまま適用すると $IP_{1\text{dB}} = OP_{1\text{dB}} - G$ となります。圧縮点での実際の利得が $G - 1$ dB であることを厳密に考慮すれば、正しくは $OP_{1\text{dB}} - G + 1 = 8$ dBm ですが、ここでは簡略化した慣例に従い 1 を加えません。
(b) An ideal linear amplifier would produce $7 + 15 = 22$ dBm at that input. The actual output is 1 dB below: $22 - 1 = 21$ dBm. Wait — the datasheet says $OP_{1\text{dB}} = 22$ dBm. This means the ideal output at the compression input would be $22 + 1 = 23$ dBm, so $IP_{1\text{dB}} = 23 - 15 = 8$ dBm.
Convention note: $OP_{1\text{dB}}$ is the actual compressed output, which is 1 dB below the ideal. So ideal output = 23 dBm; actual = 22 dBm at $P_{in} = 8$ dBm.
Starting from the Taylor series with a two-tone input $v_i = V_0(\cos\omega_1 t + \cos\omega_2 t)$, expand each power and apply the product-to-sum identities (with $\cos^2 x = \tfrac{1}{2}(1+\cos 2x)$, $\cos x\cos y = \tfrac{1}{2}[\cos(x-y)+\cos(x+y)]$, $\cos^3 x = \tfrac{3}{4}\cos x + \tfrac{1}{4}\cos 3x$):
$$\begin{aligned} v_o &= a_0 + a_1 V_0(\cos\omega_1 t + \cos\omega_2 t) + a_2 V_0^2(\cos\omega_1 t + \cos\omega_2 t)^2 \\ &\quad + a_3 V_0^3(\cos\omega_1 t + \cos\omega_2 t)^3 + \cdots \\[6pt] &= a_0 + a_1 V_0\cos\omega_1 t + a_1 V_0\cos\omega_2 t + \tfrac{1}{2}a_2 V_0^2(1+\cos 2\omega_1 t) \\ &\quad + \tfrac{1}{2}a_2 V_0^2(1+\cos 2\omega_2 t) + a_2 V_0^2\cos(\omega_1-\omega_2)t + a_2 V_0^2\cos(\omega_1+\omega_2)t \\ &\quad + a_3 V_0^3\!\left(\tfrac{3}{4}\cos\omega_1 t + \tfrac{1}{4}\cos 3\omega_1 t\right) + a_3 V_0^3\!\left(\tfrac{3}{4}\cos\omega_2 t + \tfrac{1}{4}\cos 3\omega_2 t\right) \\ &\quad + a_3 V_0^3\!\left[\tfrac{3}{2}\cos\omega_2 t + \tfrac{3}{4}\cos(2\omega_1-\omega_2)t + \tfrac{3}{4}\cos(2\omega_1+\omega_2)t\right] \\ &\quad + a_3 V_0^3\!\left[\tfrac{3}{2}\cos\omega_1 t + \tfrac{3}{4}\cos(2\omega_2-\omega_1)t + \tfrac{3}{4}\cos(2\omega_2+\omega_1)t\right] + \cdots \end{aligned}$$Group the output products by frequency:
| Frequency | Coefficient | Origin / comment |
|---|---|---|
| DC | $a_0 + a_2 V_0^2$ | $a_2$ rectifies → DC offset |
| $\omega_1,\ \omega_2$ (fundamentals) | $\bigl(a_1 V_0 + \tfrac{9}{4}a_3 V_0^3\bigr)$ | $\tfrac{3}{4}+\tfrac{3}{2}=\tfrac{9}{4}$ — the $\tfrac{3}{4}a_3 V_0^3$ term is what causes gain compression |
| $2\omega_1,\ 2\omega_2$ (2nd harmonics) | $\tfrac{1}{2}a_2 V_0^2$ | far out of band — easy to filter |
| $\omega_1\pm\omega_2$ (2nd-order intermod) | $a_2 V_0^2$ | out of band for narrow-band RF — easy to filter |
| $3\omega_1,\ 3\omega_2$ (3rd harmonics) | $\tfrac{1}{4}a_3 V_0^3$ | far out of band — easy to filter |
| $2\omega_1-\omega_2,\ 2\omega_2-\omega_1$ (IM3) | $\tfrac{3}{4}a_3 V_0^3$ | in-band — cannot be filtered. This is the IM3 product. |
| $2\omega_1+\omega_2,\ 2\omega_2+\omega_1$ | $\tfrac{3}{4}a_3 V_0^3$ | far out of band — easy to filter |
The IM3 amplitude:
$$\boxed{\,|v_{\text{IM3}}|=\tfrac{3}{4}\,|a_3|\,V_0^3\,}$$Two facts make IM3 the dominant linearity worry:
Every other 3rd-order product either falls out-of-band (filterable) or is mathematically identical in amplitude but spectrally harmless. That is why IM3 — and its figure of merit IIP3 — gets the spotlight in receiver linearity specs.
Two-tone input: $v_i = V_0(\cos\omega_1 t + \cos\omega_2 t)$. The Taylor expansion generates products at $m\omega_1 + n\omega_2$; the order is $|m|+|n|$.
Output spectrum of two-tone intermodulation products:
Problem: Two signals at $f_1 = 1000$ MHz and $f_2 = 1001$ MHz are applied to a nonlinear amplifier. List the frequencies of all IM3 products and identify which ones fall in-band.
IM3 products: $3f_1 = 3000$, $3f_2 = 3003$, $2f_1+f_2 = 3001$, $f_1+2f_2 = 3002$ — all far from passband.
In-band IM3: $2f_1-f_2 = 2000-1001 = 999$ MHz, $2f_2-f_1 = 2002-1000 = 1002$ MHz.
These fall only 1 MHz away from the fundamentals — directly in the passband and impossible to filter!
Setup — two-tone test. Drive the device with two equal-amplitude carriers close in frequency:
$$v_i = V_0\bigl[\cos\omega_1 t + \cos\omega_2 t\bigr]$$The cubic Taylor term $a_3 v_i^3$ produces every combination $\pm m\omega_1 \pm n\omega_2$ with $m+n\le 3$. Most fall far outside the RF band and are removed by a filter — the dangerous ones are $2\omega_1-\omega_2$ and $2\omega_2-\omega_1$, sitting symmetrically just outside $\omega_1$ and $\omega_2$. They land inside the RF band, where no filter can help.
Amplitude — the 3:1 slope. The IM3 component at $2\omega_1-\omega_2$ has amplitude
$$|v_{\text{IM3}}|=\tfrac{3}{4}|a_3|\,V_0^3$$The wanted output grows as $V_0$, but IM3 grows as $V_0^3$. On a log-log scale that's a 1:1 line for the fundamental and a 3:1 line for IM3 — every 1 dB more drive gives 3 dB more IM3, so spurious-free dynamic range gets worse by 2 dB per dB of drive.
Intercept point. Extrapolate(外推)both lines until they cross. The crossing input level is Extrapolate both lines until they cross. The crossing input level is 2 本の直線を交わるまで外挿します。交点の入力レベルが IIP3; the crossing output is OIP3 = IIP3 + Gain. The device never actually reaches IP3 (it compresses long before), but the imaginary intercept is a clean linearity figure of merit. Rule of thumb:
$$\mathrm{OIP3}\;\approx\;P_{1\mathrm{dB}}+10\ \text{dB}$$Why IM3 is the key linearity spec — not IM2 or IM5:
Cascade rule (input-referred). For a chain of stages with individual IIP3:
$$\frac{1}{\mathrm{IIP3}_{\text{tot}}}=\frac{1}{\mathrm{IIP3}_1}+\frac{G_1}{\mathrm{IIP3}_2}+\frac{G_1 G_2}{\mathrm{IIP3}_3}+\cdots$$Opposite of Friis noise figure: in noise the first stage dominates; in IIP3 the last stage dominates because the signal is largest there. So the LNA sets the noise floor; the mixer or PA sets the linearity ceiling.
➤As the input voltage $V_0$ increases, the voltage associated with the third-order products increases as $V_0^{\,3}$. Since power is proportional to the square of the voltage, the output power of the third-order products must increase as the cube of the input power. So for small input powers the third-order products must be very small, but will increase quickly as input power increases.
➤On a log-log ($P_{ ext{out}}$ vs $P_{ ext{in}}$ in dBm) scale the linear (1st-order) curve has slope 1 and the IM3 curve has slope 3; both compress at high drive. Extending the two idealized lines, they intersect at the hypothetical point where the 1st- and 3rd-order powers would be equal — that point is the third-order intercept point (IP3).
Problem: Research and compare Adjacent Channel Power Ratio (ACPR) and Intermodulation Ratio (IMR) as measures of amplifier/system linearity. Explain: (a) the definition of each, (b) how they differ in test signal and measurement method, (c) when each is preferred, and (d) the approximate relationship between them.
(a) Definitions:
IMR (Intermodulation Ratio) — also called IMD or IM3 — is measured using a two-tone CW test. It is defined as the ratio of the output power of a fundamental tone to the power of the third-order intermodulation product (e.g., $2\omega_1-\omega_2$):
$$\text{IMR} = \frac{P_{\omega_1}}{P_{2\omega_1-\omega_2}} \quad\text{(dBc, always positive)}$$
From the IP3 analysis: $\text{IMR (dB)} = 2[\text{OIP}_3\text{(dBm)} - P_{out}\text{(dBm)}]$
ACPR (Adjacent Channel Power Ratio) is measured using a modulated wideband signal (e.g., W-CDMA, LTE, 5G NR). It is defined as the ratio of power leaked into an adjacent channel to the total power in the main channel:
$$\text{ACPR} = \frac{P_{\text{adjacent channel}}}{P_{\text{main channel}}} \quad\text{(dBc, negative value = better)}$$
(b) Key differences:
| Feature | IMR (Two-Tone) | ACPR (Modulated Signal) |
|---|---|---|
| Test signal | Two CW tones (simple) | Digitally modulated signal (realistic) |
| What it captures | Only 3rd-order products at specific frequencies | All spectral regrowth including 3rd, 5th, 7th order and memory effects |
| Ease of measurement | Simple spectrum analyzer | Requires signal generator with IQ modulation + spectrum analyzer |
| Accuracy for real signals | Approximation only | Directly represents real-world interference |
| Relates to | IP3 (directly calculable) | IP3 (approximately, but also depends on signal statistics) |
(c) When each is preferred:
IMR/IP3 is preferred for: component-level characterization (LNAs, mixers), quick bench testing, comparing datasheets, and cascaded system analysis (the Friis-like IP3 cascade formula uses two-tone IP3).
ACPR is preferred for: system-level compliance testing (3GPP, IEEE standards mandate ACPR specs), PA characterization under realistic signal conditions, and evaluating digital predistortion (DPD) effectiveness. Modern standards (LTE, 5G NR, W-CDMA) all specify ACPR limits.
(d) Approximate relationship:
For a memoryless weakly nonlinear system with $N$ equally-spaced tones of equal power (approximating a modulated signal), analysis shows:
$$\text{ACPR (dBc)} \approx \text{IMR}_2\text{(dBc)} - C(N) \text{ dB}$$
where $\text{IMR}_2$ is the two-tone IMR, and $C(N)$ is a correction factor that depends on the number of tones. For large $N$ (good approximation of a continuous spectrum), the ACPR is approximately 6 dB worse (higher leakage) than the two-tone IMR would predict, because many intermodulation products from different tone pairs accumulate in the adjacent channel.
Rule of thumb: $\text{ACPR} \approx \text{IMR}_2 + 6$ dB (i.e., ACPR is about 6 dB worse than two-tone IMR for the same output power backoff from IP3).
Bottom line: IP3/IMR is a convenient single-number metric, but ACPR gives a more realistic picture of linearity under real modulated signals. Always measure ACPR for final system compliance.
On a dBm-vs-dBm plot: fundamental has slope 1, IM3 has slope 3. Their idealized extensions intersect at IP3.
P_IM3(dBm)P_IM3 (dBm)P_IM3(dBm) — 在某個操作點下,三階互調失真產物的 — at a given operating point, the — ある動作点における、三次相互変調積の實際輸出功率actual output power実際の出力電力。隨驅動電平而改變:P_in 上升 1 dB,P_IM3 上升 of the third-order intermodulation products. It changes with drive level: raise P_in by 1 dB and P_IM3 rises by です。駆動レベルとともに変化し、P_in が 1 dB 上がると P_IM3 は 3 dB(斜率 = 3)。可在頻譜分析儀上量測到的真實功率。(slope = 3). It is a real power you can measure on a spectrum analyser.上がります(傾き = 3)。スペクトラムアナライザで実測できる本物の電力です。
OIP3 — 固定的元件 — a fixed — 素子固有の線性度指標linearity figure of merit線形性の指標。是外推的基頻直線(斜率 1)與 IM3 直線(斜率 3)的假想交點。元件在到達 OIP3 之前就已壓縮,無法真正達到此功率。 for the device. It is the notional intersection of the extrapolated fundamental line (slope 1) with the IM3 line (slope 3). The device compresses long before it reaches OIP3, so that power can never actually be delivered.です。外挿した基本波の直線(傾き 1)と IM3 の直線(傾き 3)との仮想的な交点であり、素子は OIP3 に達するはるか手前で圧縮するため、この電力に実際に到達することはできません。不隨驅動電平改變It does not change with drive level駆動レベルによって変化しません,是元件特性值。; it is a property of the device.。素子固有の特性値です。
| P_IM3 | OIP3 | |
|---|---|---|
| 本質Nature本質 | 實際失真功率Actual distortion power実際の歪み電力 | 外推的假想交點Notional extrapolated intercept外挿による仮想的な交点 |
| 隨 P_in 改變?Changes with P_in?P_in で変化するか? | 是(斜率 3)Yes (slope 3)する(傾き 3) | 否(固定值)No (a fixed value)しない(固定値) |
| 在圖上On the plot図の上では | 斜率 3 曲線上的一點A point on the slope-3 curve傾き 3 の直線上の一点 | 固定的 y 座標A fixed y-coordinate固定された y 座標 |
| 用途Use用途 | 當前失真量The distortion right now現時点の歪み量 | 線性度規格 / 比較基準A linearity specification / basis for comparison線形性の仕様/比較の基準 |
一句話:In one sentence:一言でいえば:OIP3 是 P_IM3 在完全線性假設下「會到達」的點——是標準值;P_IM3 才是實際輸出的失真量。 OIP3 is the point P_IM3 “would reach” if everything stayed perfectly linear — a reference figure; P_IM3 is the distortion actually coming out. OIP3 はすべてが完全に線形のままなら P_IM3 が「到達するはず」の点 — 基準値です。P_IM3 こそが実際に出てくる歪み量です。
縮寫說明:Abbreviations:略語: IIP = Input Intercept Point(輸入截距點);OIP = oint (input intercept point); OIP = oint(入力インターセプト点)。OIP = Output Intercept Point(輸出截距點)。下標 3 代表三階。oint (output intercept point). The subscript 3 denotes third order.oint(出力インターセプト点)。添字 3 は三次を表します。
從 dBm 圖上的兩條直線出發。Start from the two straight lines on the dBm plot.dBm 図上の 2 本の直線から出発します。
利用截距定義求 $c$。Use the definition of the intercept to find $c$.インターセプトの定義を使って $c$ を求めます。 IIP3 定義為兩條外推線的交點輸入功率。在 $P_{in} = \text{IIP}_3$ 時,兩線都等於 OIP3: IIP3 is defined as the input power at which the two extrapolated lines cross. At $P_{in} = \text{IIP}_3$, both lines equal OIP3: IIP3 は外挿した 2 本の直線が交わる入力電力として定義されます。$P_{in} = \text{IIP}_3$ のとき、両方の直線は OIP3 に等しくなります:
$$\text{OIP}_3 = 3\,\text{IIP}_3 + c \;\Rightarrow\; c = \text{OIP}_3 - 3\,\text{IIP}_3$$代回:Substituting back:代入し直すと:
$$P_{IM3,out} = 3\,P_{in} + \text{OIP}_3 - 3\,\text{IIP}_3 \;\stackrel{\text{OIP}_3=\text{IIP}_3+G}{=}\; 3\,P_{in} + \text{IIP}_3 + G - 3\,\text{IIP}_3 = 3\,P_{in} - 2\,\text{IIP}_3 + G$$輸入端等效 IM3IM3 referred to the input入力換算した IM3(減去增益 $G$,折算回輸入):(subtract the gain $G$ to refer it back to the input):(利得 $G$ を引いて入力側へ換算します):
$$P_{IM3}\text{(dBm)} = P_{IM3,out} - G = 3\,P_{in} - 2\,\text{IIP}_3 \;\checkmark$$OIP3 = IIP3 + G。OIP3 = IIP3 + G.OIP3 = IIP3 + G。 IIP3 與 OIP3 是同一個截距點,分別從輸入軸和輸出軸讀出。把 $P_{in} = \text{IIP}_3$ 代入基頻輸出線 $P_{out} = P_{in} + G$,直接得到輸出截距: IIP3 and OIP3 are the same intercept point, read off the input axis and the output axis respectively. Substituting $P_{in} = \text{IIP}_3$ into the fundamental output line $P_{out} = P_{in} + G$ gives the output intercept directly: IIP3 と OIP3 は同じ交点を入力軸と出力軸のそれぞれから読んだものです。基本波出力の直線 $P_{out} = P_{in} + G$ に $P_{in} = \text{IIP}_3$ を代入すれば、出力インターセプトが直接得られます:
$$\text{OIP}_3 = \underbrace{\text{IIP}_3}_{P_{in}\text{ at intercept}} + G \;\checkmark$$圖中的 GThe G in the figure図中の G = 小訊號功率增益(small-signal power gain)is the small-signal power gainは小信号電力利得です。在 Po vs Pi 圖的線性區,基頻輸出線(slope = 1)與 45° 線之間有 G dB 的垂直距離——即 P. In the linear region of the Po vs Pi plot, there is a vertical distance of G dB between the fundamental output line (slope = 1) and the 45° line — that is, the intercept at which P。Po 対 Pi 図の線形領域では、基本波出力の直線(傾き = 1)と 45° 線との間に G dB の垂直距離があります — すなわち Pin = 0 時 P = 0 gives P = 0 のとき Pout = G 的截距。它同時也等於 OIP3 與 IIP3 之間的縱軸差:$G = \text{OIP3} - \text{IIP3}$。圖中雖然不一定標示箭頭,但壓縮曲線與直線外推的分歧,以及 OIP3/IIP3 的相對位置,都隱含了 G 的大小。 = G. It is also equal to the vertical gap between OIP3 and IIP3: $G = \text{OIP3} - \text{IIP3}$. The figure may not mark it with an arrow, but the divergence between the compression curve and the straight-line extrapolation, and the relative positions of OIP3 and IIP3, both encode the size of G. = G となる切片です。これは OIP3 と IIP3 の縦軸方向の差にも等しく、$G = \text{OIP3} - \text{IIP3}$ です。図には矢印が示されていないこともありますが、圧縮曲線と直線外挿のずれ、および OIP3/IIP3 の相対位置には、いずれも G の大きさが織り込まれています。
注意:Note:注意:元件在實際上遠未達到 IP3 就已經壓縮了(通常 OIP3 ≈ $P_{1\text{dB}}$ + 10 dB)。IP3 是純粹的外推交點,作為線性度指標使用,並非元件真實工作點。 in practice a device compresses long before it ever reaches IP3 (typically OIP3 ≈ $P_{1\text{dB}}$ + 10 dB). IP3 is a purely extrapolated intercept used as a linearity figure of merit, not a real operating point of the device. 実際には、素子は IP3 に達するはるか手前で圧縮します(通常 OIP3 ≈ $P_{1\text{dB}}$ + 10 dB)。IP3 は線形性の指標として使われる純粋な外挿交点であり、素子の実際の動作点ではありません。
Problem: An amplifier has $G = 12$ dB and $\text{OIP}_3 = 30$ dBm. Two equal-power tones at $P_{in} = -20$ dBm each are applied. Find (a) IIP3, (b) the output power of each fundamental, and (c) the output IM3 power.
(a) $\text{IIP}_3 = \text{OIP}_3 - G = 30 - 12 = 18$ dBm
(b) $P_{out,fund} = P_{in} + G = -20 + 12 = -8$ dBm
(c) $P_{IM3,out} = 3\,P_{out} - 2\,\text{OIP}_3 = 3(-8) - 2(30) = -24 - 60 = -84$ dBm
Alternatively: $P_{IM3,in} = 3\,P_{in} - 2\,\text{IIP}_3 = 3(-20) - 2(18) = -60 - 36 = -96$ dBm, then $P_{IM3,out} = -96 + 12 = -84$ dBm ✓
The IM3 products are 76 dB below the fundamentals.
Setup. From the single-tone Taylor expansion (keeping terms up to third order), the small-signal voltage gain at $\omega_0$ is
$$G_v(V_0) = a_1 + \tfrac{3}{4}\,a_3\,V_0^{\,2}$$Step 1 — Solve for $V_{1\text{dB}}$ at the 1 dB compression point. By definition, the gain at $V_{1\text{dB}}$ has dropped by 1 dB:
$$20\log G_{1\text{dB}} \;=\; 20\log\!\left(a_1 + \tfrac{3}{4}a_3 V_{1\text{dB}}^{\,2}\right) \;=\; 20\log a_1 - 1$$so $G_{1\text{dB}} = a_1\cdot 10^{-0.05}$. Solving the equality for $V_{1\text{dB}}$:
$$V_{1\text{dB}} \;=\; \sqrt{\frac{4\,(10^{-0.05}-1)\,a_1}{3\,a_3}}$$Step 2 — Compute $OP_{1\text{dB}}$. The output power at compression is $(G_{1\text{dB}}V_{1\text{dB}})^2 / 2Z_0$:
$$\begin{aligned} OP_{1\text{dB}} &= \frac{(G_{1\text{dB}}\,V_{1\text{dB}})^2}{2Z_0} = \frac{1}{2Z_0}\,(a_1\cdot 10^{-0.05})^2 \cdot \frac{4(1-10^{-0.05})\,|a_1|}{3\,|a_3|} \\[3pt] &= \frac{2\times 10^{-0.1}\,(1-10^{-0.05})}{3\,Z_0}\,\left|\frac{a_1^{\,3}}{a_3}\right| \quad\text{(W)} \end{aligned}$$For $Z_0 = 50\ \Omega$ and converting to dBm:
$$\boxed{\,OP_{1\text{dB}}\,(\text{dBm}) \;=\; 10\log\!\left|\frac{a_1^{\,3}}{a_3}\right| + 0.61\,}$$Step 3 — In the linear region, find $\text{OIP}_3$. The fundamental and IM3 output powers are
$$P_{\omega_1} = P_{\omega_2} = \frac{a_1^{\,2}\,V_0^{\,2}}{2Z_0},\qquad P_{2\omega_1-\omega_2} = P_{2\omega_2-\omega_1} = \frac{1}{2Z_0}\left(\tfrac{3}{4}a_3\,V_0^{\,3}\right)^2 = \frac{9\,a_3^{\,2}\,V_0^{\,6}}{32\,Z_0}$$At the intercept $V_{\text{IP}_3}$, the two are equal: $P_{\omega_1}(V_{\text{IP}_3}) = P_{2\omega_1-\omega_2}(V_{\text{IP}_3})$. Solving:
$$V_{\text{IP}_3} \;=\; \sqrt{\frac{4\,a_1}{3\,|a_3|}}$$Substituting back gives the output intercept power:
$$\text{OIP}_3 \;=\; \frac{a_1^{\,2}\,V_{\text{IP}_3}^{\,2}}{2Z_0} \;=\; \frac{2\,a_1^{\,3}}{3\,Z_0\,|a_3|}\quad\text{(W)}$$For $Z_0 = 50\ \Omega$ in dBm:
$$\boxed{\,\text{OIP}_3\,(\text{dBm}) \;=\; 10\log\!\left|\frac{a_1^{\,3}}{a_3}\right| + 11.25\,}$$Step 4 — Subtract. Both expressions share the same $10\log|a_1^{\,3}/a_3|$ term, so
$$\boxed{\,\text{OIP}_3\,(\text{dBm}) - OP_{1\text{dB}}\,(\text{dBm}) \;=\; 11.25 - 0.61 \;=\; 10.64\ \text{dB}\,}$$Bonus — the 3:1 slope rule for IM3. Take the ratio
$$\frac{P_{2\omega_1-\omega_2}}{(P_{\omega_1})^3} \;=\; \frac{9\,Z_0^{\,2}\,a_3^{\,2}}{4\,a_1^{\,6}} \;=\; \left(\frac{1}{\text{OIP}_3}\right)^{\!2}$$which gives the textbook IM3-from-fundamental formula in dB:
$$\boxed{\,P_{2\omega_1-\omega_2}\,(\text{dBm}) \;=\; 3\,P_{\omega_1}\,(\text{dBm}) - 2\,\text{OIP}_3\,(\text{dBm})\,}$$Every 1 dB rise in $P_{\omega_1}$ adds 3 dB to the IM3 product, so the gap shrinks by 2 dB per dB of drive — that is the dynamic-range squeeze you fight with cascaded IIP3 budgeting.
$G=10$ dB, $F=5$ dB, $\text{OIP}_3 = 20$ dBm, $T_0=290$ K, $B=1$ MHz.
$OP_{1\text{dB}} = 20 - 10.64 = 9.36$ dBm
$N_o = -99$ dBm. $OP_{sf} = \frac{2}{3}(9.36) + 7.09 + \frac{1}{3}(-99) = -19.67$ dBm
Problem: A mixer has $\text{IIP}_3 = +5$ dBm and conversion loss $L = 7$ dB. Estimate $IP_{1\text{dB}}$ and $OP_{1\text{dB}}$.
$\text{OIP}_3 = \text{IIP}_3 + G = 5 + (-7) = -2$ dBm
$OP_{1\text{dB}} = \text{OIP}_3 - 10.64 = -2 - 10.64 = -12.64$ dBm
$IP_{1\text{dB}} = OP_{1\text{dB}} - G = -12.64 - (-7) = -5.64$ dBm
(Rule of thumb check: $\text{IIP}_3 - IP_{1\text{dB}} \approx 5 - (-5.6) = 10.6$ dB ✓)
➤A power range limited at the low end by noise and at the high end by the compression point. This is usually considered the usable power range for a linear power amplifier.
➤A power range limited at the low end by the third-order intermodulation product equal to the noise level and at the high end by the corresponding output signal power. This is usually considered the usable power range for low-noise amplifiers or mixers in a receiver.
$P_{\text{out}}$ vs $P_{\text{in}}$ — the two dynamic ranges $DR_l$ (linear) and $DR_f$ (spurious-free) sit between the noise floor $N_o$ and the compression / IM3 corners:
Setup. From the 3:1 IM3 slope rule (§3.5), the fundamental output power can be written in terms of IM3 and OIP3:
$$P_{\omega_1} \;=\; \bigl(P_{2\omega_1-\omega_2}\bigr)^{\!1/3}\,(\text{OIP}_3)^{2/3}$$Step 1 — $DR_f$ in linear form. Substitute into the $DR_f$ definition with the constraint $P_{2\omega_1-\omega_2}=N_o$:
$$DR_f \;=\; \left.\frac{P_{\omega_1}}{P_{2\omega_1-\omega_2}}\right|_{P_{2\omega_1-\omega_2}=N_o} \;=\; \left.\frac{(P_{2\omega_1-\omega_2})^{1/3}(\text{OIP}_3)^{2/3}}{P_{2\omega_1-\omega_2}}\right|_{P_{2\omega_1-\omega_2}=N_o} \;=\; \left(\frac{\text{OIP}_3}{N_o}\right)^{\!2/3} \;=\; \left(\frac{\text{OIP}_3}{kT_0\,B\cdot G_A\cdot F}\right)^{\!2/3}$$In dB:
$$\boxed{\,DR_f\,(\text{dB}) \;=\; \tfrac{2}{3}\,\text{OIP}_3\,(\text{dBm}) - \tfrac{2}{3}\,N_o\,(\text{dBm})\,}$$Step 2 — $DR_l$ in dB form. Directly from the definition $DR_l = OP_{1\text{dB}}/N_o$:
$$\boxed{\,DR_l\,(\text{dB}) \;=\; OP_{1\text{dB}}\,(\text{dBm}) - N_o\,(\text{dBm})\,}$$Step 3 — relate $DR_l$ and $DR_f$. Solving Step 1 for $\text{OIP}_3$ and substituting into Step 2:
$$DR_l\,(\text{dB}) \;=\; \tfrac{3}{2}\,DR_f\,(\text{dB}) + OP_{1\text{dB}}\,(\text{dBm}) - \text{OIP}_3\,(\text{dBm})$$Using §3.5's empirical rule $\text{OIP}_3 - OP_{1\text{dB}} = 10.64$ dB:
$$\boxed{\,DR_l\,(\text{dB}) \;=\; \tfrac{3}{2}\,DR_f\,(\text{dB}) - 10.64\ \text{dB}\,}$$Comparing the two ranges:
Step 4 — output spurious-free power $OP_{sf}$. The fundamental output level at which the IM3 product equals the noise floor:
$$\begin{aligned} OP_{sf}\,(\text{dBm}) &\equiv \left.P_{\omega_1}\right|_{P_{2\omega_1-\omega_2}=N_o} \;=\; DR_f\,(\text{dB}) + N_o\,(\text{dBm}) \\[3pt] &=\; \tfrac{2}{3}\,\text{OIP}_3\,(\text{dBm}) + \tfrac{1}{3}\,N_o\,(\text{dBm}) \\[3pt] &=\; \tfrac{2}{3}\,OP_{1\text{dB}}\,(\text{dBm}) + 7.09\,(\text{dBm}) + \tfrac{1}{3}\,N_o\,(\text{dBm}) \end{aligned}$$The constant $7.09 = \tfrac{2}{3}\times 10.64$ dB carries the OIP3-vs-P1dB offset into the spur-free output expression.
With minimum detectable SNR specified: $DR_f = \tfrac{2}{3}[\text{OIP}_3 - N_o] - \text{SNR}_{min}$.
Always: $DR_l - DR_f \approx 21.3$ dB.
$F=7$ dB, $OP_{1\text{dB}}=25$ dBm, $G=40$ dB, $\text{OIP}_3=35$ dBm, $T_a=150$ K, SNR$_{min}$=10 dB, $B=100$ MHz.
$N_o = GkB[T_a+(F-1)T_0] = 10^4\times1.38\times10^{-23}\times10^8\times[150+4.01\times290] = -47.4$ dBm
$DR_l = 25 - (-47.4) = 72.4$ dB
$DR_f = \frac{2}{3}[35-(-47.4)] - 10 = \frac{2}{3}(82.4) - 10 = 54.9 - 10 = 44.9$ dB
Problem: A receiver has $\text{OIP}_3 = 25$ dBm, $G = 30$ dB, $F = 4$ dB, $B = 10$ MHz, $T_0 = 290$ K. Find $DR_l$ and $DR_f$ (no minimum SNR constraint).
$N_o = kT_0B\cdot G\cdot F$ (since source is at $T_0$)
$= 1.38\times10^{-23}\times290\times10^7\times10^3\times10^{0.4}$
$= 4\times10^{-14}\times1000\times2.512 = 1.005\times10^{-10}$ W $= -70.0$ dBm
$OP_{1\text{dB}} = \text{OIP}_3 - 10.64 = 25 - 10.64 = 14.36$ dBm
$DR_l = 14.36 - (-70.0) = 84.4$ dB
$DR_f = \frac{2}{3}[25 - (-70.0)] = \frac{2}{3}\times95 = 63.3$ dB
Check: $DR_l - DR_f = 84.4 - 63.3 = 21.1$ dB $\approx 21.3$ dB ✓
Two-stage cascade and its equivalent single-block representation:
At the output of the first stage:
$$P'_{2\omega_1-\omega_2} \;=\; \frac{(P'_{\omega_1})^3}{(\text{OIP}_3')^2}\;\Longrightarrow\;V'_{2\omega_1-\omega_2} \;=\; \sqrt{2 Z_0\cdot P'_{2\omega_1-\omega_2}} \;=\; \frac{\sqrt{2 Z_0\,(P'_{\omega_1})^3}}{\text{OIP}_3'}$$At the output of the second stage:
$$V''_{2\omega_1-\omega_2} \;=\; \sqrt{G_2}\cdot V'_{2\omega_1-\omega_2} \;+\; \frac{\sqrt{2 Z_0\,(P''_{\omega_1})^3}}{\text{OIP}_3''} \;=\; \frac{\sqrt{2 G_2 Z_0\,(P'_{\omega_1})^3}}{\text{OIP}_3'} \;+\; \frac{\sqrt{2 Z_0\,(P''_{\omega_1})^3}}{\text{OIP}_3''}$$Since $P''_{\omega_1} = G_2\,P'_{\omega_1}$:
$$V''_{2\omega_1-\omega_2} \;=\; \frac{\sqrt{2 G_2 Z_0\,(P''_{\omega_1}/G_2)^3}}{\text{OIP}_3'} \;+\; \frac{\sqrt{2 Z_0\,(P''_{\omega_1})^3}}{\text{OIP}_3''} \;=\; \left(\frac{1}{G_2\cdot\text{OIP}_3'} + \frac{1}{\text{OIP}_3''}\right)\sqrt{2 Z_0\,(P''_{\omega_1})^3}$$ $$P''_{2\omega_1-\omega_2} \;=\; \frac{(V''_{2\omega_1-\omega_2})^2}{2Z_0} \;=\; \left(\frac{1}{G_2\cdot\text{OIP}_3'} + \frac{1}{\text{OIP}_3''}\right)^{\!2}\!(P''_{\omega_1})^3 \;\equiv\; \frac{(P''_{\omega_1})^3}{(\text{OIP}_3)^2}$$where $\text{OIP}_3$ is the equivalent output third-order intercept point of the cascaded network.
$$\Longrightarrow\quad \frac{1}{\text{OIP}_3} \;=\; \frac{1}{G_2\cdot\text{OIP}_3'} \;+\; \frac{1}{\text{OIP}_3''}$$In general, for $N$ networks that are cascaded:
$$\frac{1}{\text{OIP}_3} \;=\; \frac{1}{G_2 G_3\cdots G_N\cdot\text{OIP}_3^{(1)}} \;+\; \frac{1}{G_3\cdots G_N\cdot\text{OIP}_3^{(2)}} \;+\; \cdots \;+\; \frac{1}{G_N\cdot\text{OIP}_3^{(N-1)}} \;+\; \frac{1}{\text{OIP}_3^{(N)}}$$With overall gain $G = G_1 G_2 \cdots G_{N-1}\,G_N$:
$$\frac{1}{\text{OIP}_3} \;=\; \frac{1}{G}\!\left(\frac{1}{\text{OIP}_3^{(1)}/G_1} + \frac{1}{\text{OIP}_3^{(2)}/(G_1 G_2)} + \cdots + \frac{1}{\text{OIP}_3^{(N-1)}/(G_1 G_2\cdots G_{N-1})} + \frac{1}{\text{OIP}_3^{(N)}/(G_1 G_2\cdots G_{N-1}\,G_N)}\right)$$The above expression shows that the third-order intercept point of a cascaded system is dominated by the last few stages, since the later stages divided by a larger product of the gains reduce $\text{IP}_3$ more significantly than the earlier stages.
Noise figure: first stage dominates (later stages divided by preceding gain).
IP3: last stage dominates (later stages multiplied by preceding gain).
This creates a fundamental design tension in receiver chains.
Amplifier: $G=20$ dB, $\text{OIP}'_3=22$ dBm $=158$ mW. Mixer: $L=6$ dB ($G_2=0.25$), $\text{IIP}''_3=13$ dBm → $\text{OIP}''_3 = 7$ dBm $=5$ mW.
$\frac{1}{\text{OIP}_3} = \frac{1}{0.25\times158} + \frac{1}{5} = \frac{1}{39.5} + \frac{1}{5} = 0.0253 + 0.200 = 0.2253$
$\text{OIP}_3 = 4.4$ mW $= 6.4$ dBm. (Dominated by the mixer's OIP3!)
Problem: Three cascaded stages: LNA ($G_1=15$ dB, $\text{OIP}_3^{(1)}=25$ dBm), Filter ($G_2=-2$ dB, $\text{OIP}_3^{(2)}=40$ dBm), Mixer ($G_3=-8$ dB, $\text{OIP}_3^{(3)}=10$ dBm). Find the overall OIP3.
Convert: $G_1=31.62$, $G_2=0.631$, $G_3=0.158$. $G_{total} = 31.62\times0.631\times0.158 = 3.15 = 4.98$ dB.
$\text{OIP}_3^{(1)}=316.2$ mW, $\text{OIP}_3^{(2)}=10{,}000$ mW, $\text{OIP}_3^{(3)}=10$ mW.
Using the formula (all powers in mW, referred to overall output):
$\frac{1}{\text{OIP}_3} = \frac{1}{\text{OIP}_3^{(1)}\cdot G_2 G_3} + \frac{1}{\text{OIP}_3^{(2)}\cdot G_3} + \frac{1}{\text{OIP}_3^{(3)}}$
$= \frac{1}{316.2\times0.631\times0.158} + \frac{1}{10000\times0.158} + \frac{1}{10}$
$= \frac{1}{31.52} + \frac{1}{1580} + \frac{1}{10} = 0.0317 + 0.000633 + 0.100 = 0.1324$
$\text{OIP}_3 = 7.55$ mW $= 8.78$ dBm
(Dominated by the mixer — the last-stage term $1/10 = 0.100$ is the largest contributor.)
➤Intermodulation products can also be generated by passive nonlinear effects — anywhere there is a metal-to-metal contact (connectors, cables, antennas, etc.). The result is called passive intermodulation (PIM).
A problem bank covering every topic in this chapter — noise sources & thermal noise, PSD & autocorrelation, modulation/BER, noise figure & temperature, Friis cascade, antenna noise, link budgets, and nonlinear distortion (P1dB, IP3, dynamic range). Click ? to flag a problem for review or ✓ to mark it done — state persists in your browser.
Detailed worked solutions to all 42 problems, with formulas, substitutions, unit checks, and design interpretation. Double-click any problem above to jump to its solution; double-click any solution to jump back to the question.
Thermal — thermal vibration of bound charges in any resistive element (resistors, lossy lines, atmosphere); white spectrum.
Shot — discrete charge carriers crossing a junction; $\propto I_{DC}$; dominant in diodes/transistors.
Flicker (1/f) — surface defects in semiconductors; sets oscillator phase-noise floor.
Plasma — random motion in ionized gas (atmosphere, sparking contacts).
Quantum — quantized photons; negligible at microwave because $hf \ll kT$.
(a) $v_n^s = \sqrt{4kTBR} = \sqrt{4(1.38\!\times\!10^{-23})(290)(10^7)(50)} = \sqrt{8.0\!\times\!10^{-12}} \approx 2.83\;\mu\text{V rms}$.
(b) $P_n = kTB = (1.38\!\times\!10^{-23})(290)(10^7) = 4.0\!\times\!10^{-14}\;\text{W}$.
In dBm: $10\log(4.0\!\times\!10^{-11}\;\text{mW}) = -104\;\text{dBm}$. Equivalently $-174 + 10\log(10^7) = -174+70 = -104$ dBm.
$kT_0 = (1.38\!\times\!10^{-23})(290) = 4.00\!\times\!10^{-21}$ W/Hz $= 4.00\!\times\!10^{-18}$ mW/Hz.
$10\log(4.00\!\times\!10^{-18}) = -174$ dBm/Hz. ✓
Floor $= -174 + 10\log B$:
Two-sided $S(f) = n_0/2$ for all $f$. An ideal BPF passes $[-f_c-B/2, -f_c+B/2]$ and $[f_c-B/2, f_c+B/2]$.
$P_n = \displaystyle\int_{-\infty}^{\infty} S(f)\,|H(f)|^2 df = \tfrac{n_0}{2}(B) + \tfrac{n_0}{2}(B) = n_0 B = kTB$.
The two halves account for negative and positive frequencies — both are physical (real-signal symmetry).
Method
Use the Wiener-Khinchin theorem: autocorrelation and PSD are Fourier transform pairs. For ideal white noise the PSD is constant, so its inverse Fourier transform is an impulse.
$R(\tau) = \mathcal{F}^{-1}\{S(f)\} = \mathcal{F}^{-1}\{n_0/2\} = (n_0/2)\,\delta(\tau)$.
For $\tau \neq 0$: $R(\tau)=E[n(t)n(t+\tau)] = 0$ — distinct samples are uncorrelated.
At $\tau = 0$: $R(0) = \int S(f)\,df = \infty$ — infinite total power. White noise is a mathematical idealization; real noise is band-limited.
Method
Start from the finite-energy Parseval relation, then use the power-signal version for stationary random noise.
For finite-energy signals: $\int|x(t)|^2 dt = \int|X(f)|^2 df$.
For random/power signals, $S(f)$ is the time-averaged $|X_T(f)|^2/T$ as $T\to\infty$.
Setting $\tau=0$: $R(0) = E[|n(t)|^2]$ — the mean-square value, i.e. the time-average power.
Wiener–Khinchin: $R(0) = \int S(f)\,df$. Hence $\int S(f)\,df$ is the total power.
For a receiver filter, include the filter response: $P_n=\int S(f)|H(f)|^2df$. With an ideal bandpass of physical bandwidth $B$ and two-sided $S(f)=n_0/2$, the positive and negative passbands contribute $n_0B/2+n_0B/2=n_0B$.
Coherent demod: multiply by $2\cos\omega_c t$, lowpass.
SSB: Signal occupies one sideband only. Multiplication recovers baseband; noise in that single sideband is shifted to baseband. Both signal power and noise power transfer 1:1 → SNR$_{out}$ = SNR$_{in}$.
DSB-SC: Signal in both sidebands adds coherently after demod (signal voltage doubles → power $\times 4$). Noise in upper and lower sidebands is uncorrelated (white) so noise powers add (noise $\times 2$). Net SNR gain $= 4/2 = 2 = +3$ dB.
Solve $P_e = \tfrac12\mathrm{erfc}(\sqrt{E_b/n_0})$:
Input: $\pm V\cos\omega_c t + n(t)$ over $[0,T]$. Multiply by $\cos\omega_c t$, integrate.
(a) Signal: $\int_0^T (\pm V)\cos^2\omega_c t\,dt = \pm V T/2$ (peak $\omega_c T \gg 1$ assumption).
(b) Noise variance: $\sigma^2 = E\!\left[\left(\!\int_0^T n(t)\cos\omega_c t\,dt\right)^{\!2}\right] = \int_0^T\!\!\int_0^T R_n(t-s)\cos\omega_c t\cos\omega_c s\,ds\,dt$.
With $R_n(\tau)=(n_0/2)\delta(\tau)$: $\sigma^2 = (n_0/2)\int_0^T\cos^2\omega_c t\,dt = n_0 T/4$.
Some books use the convention $\sigma^2_{out} = n_0/T$ after normalizing by $T$.
(c) SNR: Signal power $(VT/2)^2 = V^2T^2/4$; noise power $n_0 T/4$. SNR $= V^2 T / n_0 = 2E_b/n_0$ where $E_b = V^2T/2$.
$\eta = m^2/(2+m^2)$:
$E_b/n_0$ (dB) at BER $= 10^{-5}$:
$\eta_s = \log_2 M$ (bps/Hz, ideal Nyquist):
Need $\eta_s \ge 24/6 = 4$ bps/Hz.
$S/N|_{dB} = (E_b/n_0)|_{dB} + 10\log(R_b/B)$.
$10\log(48.6/30) = 10\log(1.62) = 2.10$ dB.
(a) For 16-QAM at $R_b/B = 12/4 = 3$:
$S/N = 14.4 + 10\log 3 = 14.4 + 4.77 = 19.2$ dB.
(b) QPSK at the same $R_b=12$ Mbps needs $B = 6$ MHz (since $\eta_s = 2$):
$S/N = 9.6 + 10\log(12/6) = 9.6 + 3.01 = 12.6$ dB.
(a) GaAs HEMT LNA at 4 GHz: thermal (Johnson-Nyquist) noise from the channel resistance dominates; shot noise of the gate is negligible because gate current is essentially zero in a HEMT.
(b) Si BJT mixer at 100 MHz, low $I_C$: shot noise of the base/emitter junctions dominates because thermal noise of the small-signal $r_e$ is comparable but the $2qI$ shot term wins at low currents and low frequency.
(c) Varactor multiplier at 8 GHz: thermal noise of the series resistance and the lossy substrate; the device is essentially passive (no DC bias current → no shot noise).
(a) $v_n^{rms} = \sqrt{4kTRB} = \sqrt{4(1.38\!\times\!10^{-23})(300)(2200)(5\!\times\!10^6)} \approx 6.0\ \mu$V.
(b) Available noise power $P_n = kTB = (1.38\!\times\!10^{-23})(300)(5\!\times\!10^6) = 2.07\!\times\!10^{-14}$ W $= -106.8$ dBm.
(c) At 77 K, $v_n^{rms}$ scales as $\sqrt{T}$: factor $\sqrt{77/300} = 0.507$, so new value $\approx 3.0\ \mu$V (and noise power drops by $10\log(77/300) = -5.9$ dB).
For an ideal first-order RC LPF, the noise bandwidth is $B_n = (\pi/2) f_c = 1.571\times f_c = 157.1$ MHz.
Total output noise voltage variance: $\sigma_v^2 = 2 S_n B_n$ (factor 2 because two-sided integrates positive + negative frequencies):
$\sigma_v^2 = 2 \times 2\!\times\!10^{-18} \times 1.571\!\times\!10^8 = 6.28\!\times\!10^{-10}$ V$^2$.
$v_{rms} = \sqrt{\sigma_v^2} = 25.1\ \mu$V across 50 $\Omega$.
Noise power into 50 $\Omega$: $P_n = v_{rms}^2 / 50 = 1.26\!\times\!10^{-11}$ W $= -79.0$ dBm.
Signal power: $S_i = (1\!\times\!10^{-3})^2 / 50 = 2\!\times\!10^{-8}$ W $= -47.0$ dBm.
Noise power across 30 kHz: $N_i = -174 + 10\log(30\!\times\!10^3) = -174 + 44.8 = -129.2$ dBm.
(a) SNR$_i = -47 - (-129.2) = 82.2$ dB.
(b) For a coherent SSB demodulator the output SNR equals the input SNR within a constant; SNR$_o = 82.2$ dB.
(c) Improvement = 0 dB. SSB is a linear demodulator — no SNR gain, but also no penalty (unlike envelope-detected AM, which has a threshold).
BPSK at $E_b/n_0 = 7$ dB: linear $E_b/n_0 = 5.01$. $P_b = Q(\sqrt{2\times 5.01}) = Q(3.17) \approx 7.6\!\times\!10^{-4}$.
64-QAM at the same $E_b/n_0$: required $E_b/n_0$ for $P_b = 10^{-6}$ is $\approx 18.5$ dB; at 7 dB the BER is essentially $> 10^{-1}$ (deep below threshold).
Quantitatively, for square M-QAM with Gray coding: $P_b \approx \frac{4}{\log_2 M}\Big(1-\frac{1}{\sqrt{M}}\Big) Q\!\left(\sqrt{\frac{3\log_2 M}{M-1}\,\frac{E_b}{n_0}}\right)$. Plug $M=64$, $E_b/n_0 = 5.01$ → argument $\sqrt{(3\cdot 6/63)\cdot 5.01} = \sqrt{1.43} = 1.20$, so $Q(1.2) \approx 0.115$ and $P_b \approx 0.115$.
(a) Matched filter (integrate-and-dump): noise bandwidth equals signal bandwidth; $P_b = Q(\sqrt{2\, E_b/n_0}) = Q(\sqrt{2\times 7.94}) = Q(3.99) \approx 3.4\!\times\!10^{-5}$.
(b) Brick-wall LPF at $B = 1.5$ MHz (excess bandwidth $50\%$): noise power is $1.5\times$ that of the matched filter, so effective $E_b/n_0$ drops by $10\log(1.5) = 1.76$ dB. New $E_b/n_0 = 9 - 1.76 = 7.24$ dB → $P_b = Q(\sqrt{2\times 5.30}) = Q(3.26) \approx 5.6\!\times\!10^{-4}$.
| Modulation | $E_b/n_0$ for $10^{-6}$ | $\eta_s$ (max bps/Hz) | Penalty vs BPSK | Used in |
|---|---|---|---|---|
| BPSK | 10.5 dB | 1 | 0 dB | satellite TT&C |
| QPSK | 10.5 dB | 2 | 0 dB | DVB-S, GPS |
| 8-PSK | 14.0 dB | 3 | +3.5 dB | DVB-S2 (some MODCODs) |
| 16-QAM | 14.4 dB | 4 | +3.9 dB | Wi-Fi, LTE, 5G NR |
| 64-QAM | 18.6 dB | 6 | +8.1 dB | Wi-Fi 5/6, LTE, 5G NR |
| 256-QAM | 23.5 dB | 8 | +13.0 dB | Wi-Fi 6/7, 5G mmWave |
(a) Symbol rate $R_s = R_b / \log_2 M = 80 / 4 = 20$ Msym/s. Spectral efficiency $\eta_s = R_b / B = 80 / 25 = 3.2$ bps/Hz.
(b) From Pozar Table 14.1 / standard curves: 16-QAM BER $= 10^{-5}$ → $E_b/n_0 \approx 13.4$ dB.
(c) $S/N = E_b/n_0 + 10\log(R_b/B) = 13.4 + 10\log 3.2 = 13.4 + 5.05 = 18.5$ dB.
(d) Noise floor $N = kT_{sys}B = (1.38\!\times\!10^{-23})(400)(25\!\times\!10^6) = 1.38\!\times\!10^{-13}$ W $= -98.6$ dBm. Required $P_r = N + (S/N) = -98.6 + 18.5 = -80.1$ dBm.
$T_e = (F-1)T_0$, $T_0 = 290$ K:
$F = 1 + T_e/T_0$:
Output noise added (source at 0 K) $= G\,k T_e B$. Solve:
$T_e = N_{add}/(GkB) = (4\!\times\!10^{-13})/(100 \cdot 1.38\!\times\!10^{-23} \cdot 10^8) = 4\!\times\!10^{-13}/1.38\!\times\!10^{-13} \approx 2.9$ K.
$F = 1 + T_e/T_0 = 1 + 2.9/290 \approx 1.010 \;\Rightarrow\; 0.04$ dB.
$T_e = (T_h - YT_c)/(Y-1) = (873 - 2.5\!\cdot\!77)/(2.5-1) = (873 - 192.5)/1.5 = 680.5/1.5$.
$T_e = 454$ K. $F = 1 + 454/290 = 2.56$.
$L = 10^{0.25} = 1.778$. At $T_p = T_0$: $F = L = 2.5$ dB exactly.
$T_e = (L-1)T_p = 0.778 \cdot 290 = 226$ K.
Why $F = L$: at $T_p = T_0$, both source noise and added noise are $kT_0B$. Power gets attenuated by $L$ but added noise refills to keep output noise at $kT_0B$ → SNR drops by exactly $L$.
Equivalently, use the temperature form: $F=1+T_e/T_0=1+(L-1)T_0/T_0=L$. This is why a passive room-temperature loss has noise figure equal to insertion loss.
$L = 10^{0.3} = 2.0$. At $T_p = 200$ K: $T_e = (L-1)T_p = 1\cdot 200 = 200$ K.
If naively $T_p = 290$: $T_e = 1\cdot 290 = 290$ K.
Linear: $F_1=1.585$, $G_1=31.62$; $F_2=5.01$, $G_2=0.251$; $F_3=2.512$.
$F = F_1 + (F_2-1)/G_1 + (F_3-1)/(G_1 G_2)$
$= 1.585 + 4.01/31.62 + 1.512/(31.62\cdot 0.251)$
$= 1.585 + 0.127 + 1.512/7.93 = 1.585 + 0.127 + 0.190 = 1.902$.
(a) Clean: $F_1=1.585\;(2\,\text{dB})$, $G_1=100$; $F_2=6.31\;(8\,\text{dB})$, $G_2=0.251$; $F_3=2.512\;(4\,\text{dB})$.
$F_a = 1.585 + 5.31/100 + 1.512/(100\cdot 0.251) = 1.585 + 0.053 + 0.060 = 1.698 \Rightarrow 2.30$ dB.
(b) With cable: add 1.5 dB cable ($F_0 = L = 1.413$, $G_0 = 0.708$) at front:
$F_b = 1.413 + (1.585-1)/0.708 + 5.31/(0.708\cdot 100) + 1.512/(0.708\cdot 100\cdot 0.251)$
$= 1.413 + 0.826 + 0.075 + 0.085 = 2.40 \Rightarrow 3.80$ dB.
A matched, lossless Wilkinson splits input power equally to two output ports — each port sees $P_{in}/2$, i.e. 3 dB lower.
From one output port's perspective, the divider looks like a 3 dB lossy two-port (input → that output), so by the lossy-line rule, $F = L = 3$ dB at $T_p = T_0$.
No contradiction: the "missing" half-power didn't dissipate — it went out the other output port. The receiver at port 2 still sees its input SNR halved.
Detailed SNR view: for one branch, the desired signal available at the receiver is halved. The source noise delivered into that branch is also halved, but the branch still sees a matched $T_0$ environment. Referred to a single output path, the available SNR is degraded by the same factor as a 3 dB attenuator.
$T_a = \eta_{rad}T_b + (1-\eta_{rad})T_p = 0.9\cdot 50 + 0.1\cdot 290 = 45 + 29 = 74$ K.
Pointed at the ground ($T_b = 290$): $T_a = 0.9\cdot 290 + 0.1\cdot 290 = 290$ K.
(a) EIRP $= P_t + G_t = 10\log(100) + 35 = 20 + 35 = 55$ dBW $= 85$ dBm.
(b) $\lambda = 3\!\times\!10^8/12\!\times\!10^9 = 0.025$ m.
FSPL $= 32.4 + 20\log(12000) + 20\log(36000) = 32.4 + 81.6 + 91.1 \approx 205$ dB.
(c) $P_r = $ EIRP $- $ FSPL $+ G_r = 55 - 205 + 30 = -120$ dBW $= -90$ dBm.
$G/T = G_r - 10\log(T_a + T_e) = 30 - 10\log(80+120) = 30 - 10\log 200 = 30 - 23.0 = 7.0$ dB/K.
With $T_e \to 60$ K: $G/T = 30 - 10\log(140) = 30 - 21.46 = 8.54$ dB/K.
$10\log(kB) = -228.6 + 10\log(2.7\!\times\!10^7) = -228.6 + 74.3 = -154.3$ dBW/Hz·Hz.
CNR $= $ EIRP $+ G/T - $ FSPL $- 10\log(kB)$
$= 55 + 12 - 205 - (-154.3) = 55 + 12 - 205 + 154.3 = 16.3$ dB.
QPSK at BER $=10^{-6}$ needs $E_b/n_0 \approx 10.5$ dB; with $R_b/B=2$ → $S/N_{req} \approx 13.5$ dB.
$F = 2$ ($3$ dB), so $(F-1)T_0 = 290$ K. Effective noise temp at antenna port $= T_a + (F-1)T_0 = 100 + 290 = 390$ K.
$S_{i,min} = kB\cdot 390 \cdot 10 = (1.38\!\times\!10^{-23})(10^6)(390)(10) = 5.4\!\times\!10^{-14}$ W $= -132.7$ dBm.
Why not $kT_0BF$? $kT_0BF \cdot (S_o/N_o) = (1.38e^{-23})(290)(10^6)(2)(10) = 8.0e^{-14}$ W $= -130.9$ dBm — about 1.8 dB too high. Because the antenna is cold ($T_a = 100$ K $\ne T_0$), the standard formula overstates the noise.
$L_T = 10^{0.15} = 1.413$.
(a) $T_{TL} = (L_T-1)T_p = 0.413\cdot 290 = 119.7$ K.
$T_{REC} = (F-1)T_0 = (1.585-1)(290) = 169.7$ K.
(b) $T_{SYS} = T_a + T_{TL} + L_T\cdot T_{REC} = 60 + 119.7 + 1.413\cdot 169.7 = 60 + 119.7 + 239.8 = 419.5$ K.
(c) Equivalent system NF: $F_{sys} = 1 + T_{SYS}/T_0 = 1 + 419.5/290 = 2.45 \Rightarrow 3.89$ dB.
$L_T = 10^{0.2} = 1.585$, $F = 10^{0.4} = 2.512$.
$T_{TL} = 0.585\cdot 270 = 158.0$ K; $T_{REC} = 1.512\cdot 290 = 438.5$ K.
$T_{SYS} = 200 + 158.0 + 1.585\cdot 438.5 = 200 + 158 + 695 = 1053$ K.
$F_{sys} = 1 + 1053/290 = 4.63 \Rightarrow 6.65$ dB.
Sanity check at $T_a = T_0 = 290$ K:
$T_{SYS} = 290 + 158 + 695 = 1143$ K, $F_{sys} = 1 + 1143/290 = 4.94 \Rightarrow 6.94$ dB.
Friis (line + receiver): $F_{line+rec} = L_T \cdot F = 1.585\cdot 2.512 = 3.98 \Rightarrow 6.0$ dB. With $T_p = 270 \neq T_0$ the equality is approximate; if we set $T_p = T_0$ also, $T_{SYS}$ would be exactly $T_0\cdot F_{tot}$.
Open-ended; one workable budget for 64-QAM, $R_b = 100$ Mbps, BER $=10^{-6}$ ($E_b/n_0 = 22.5$ dB):
(a) $F = 2.5$ dB → linear $F = 10^{0.25} = 1.778$. $T_e = (F-1)T_0 = 0.778 \times 290 = 225.6$ K.
(b) New source $T_s = 100$ K → $T_{sys} = 100 + 225.6 = 325.6$ K. Effective $F$ at this source temperature: $F_{eff} = 1 + T_e/T_s = 1 + 225.6/100 = 3.256 = 5.13$ dB.
$Y = (T_h + T_e) / (T_c + T_e) = 1.45$
$\Rightarrow T_e = (T_h - Y T_c) / (Y - 1) = (470 - 1.45 \times 80) / 0.45 = 354/0.45 = 786.7$ K.
$F = 1 + T_e/T_0 = 1 + 786.7/290 = 3.71 = 5.69$ dB.
(a) Input-referred noise: $kT_0BF = -174 + 10\log(10^7) + 4 = -174 + 70 + 4 = -100$ dBm.
(b) Output noise: $G + (-100) = 30 + (-100) = -70$ dBm.
(c) SNR$_i = -80 - (-100) = 20$ dB. Output: $S_o = -80 + 30 = -50$ dBm; SNR$_o = -50 - (-70) = 20$ dB.
(a) Passive line at $T_p = T_0$: $F_{line} = L = 2$ dB → linear $1.585$. $T_{line} = (L-1)T_p = 0.585 \times 290 = 169.7$ K.
(b) LNA: $F_{LNA} = 1.5$ dB = 1.413, $T_{LNA} = (1.413-1)\times 290 = 119.8$ K. Cascade with line first ($G_1 = 1/L = 0.631$):
$T_{cas} = T_{line} + T_{LNA}/G_1 = 169.7 + 119.8/0.631 = 169.7 + 189.8 = 359.5$ K.
$F_{cas} = 1 + 359.5/290 = 2.24 = 3.50$ dB.
(c) Sensitivity penalty: $F_{cas} - F_{LNA} = 3.50 - 1.5 = 2.0$ dB — exactly the line loss (this is the universal "every dB of pre-LNA loss costs 1 dB of NF" rule).
Linear gains/figures: $G_1 = 31.6$, $F_1 = 1.585$ (T₁ = 169.7 K); $L_2 = 3.98$, $G_2 = 1/3.98 = 0.251$, $F_2 = 6.31$ (T₂ = 1540 K); $G_3 = 1000$, $F_3 = 1.585$ (T₃ = 169.7 K).
$F_{cas} = F_1 + (F_2-1)/G_1 + (F_3-1)/(G_1 G_2) = 1.585 + 5.31/31.6 + 0.585/(31.6\times 0.251)$
$= 1.585 + 0.168 + 0.0738 = 1.827 = 2.62$ dB.
An ideal Wilkinson is lossless (3 dB split is just power division, not insertion loss). The 0.3 dB excess loss per arm is the actual insertion loss seen by signal and noise alike.
From one output port: looking back, the device behaves as a 3 dB attenuator (split) + 0.3 dB extra loss = 3.3 dB total loss at $T_p = 290$ K.
$T_e = (L - 1) T_p = (10^{0.33} - 1) \times 290 = (2.14 - 1) \times 290 = 330.5$ K.
$F = L = 3.3$ dB per arm.
(a) $T_a = \eta_{rad} T_{sky} + (1-\eta_{rad}) T_p = 0.85 \times 20 + 0.15 \times 290 = 17 + 43.5 = 60.5$ K.
(b) $T_{sys} = T_a + T_e = 60.5 + 60 = 120.5$ K.
(c) $N = kT_{sys}B = 1.38\!\times\!10^{-23} \times 120.5 \times 36\!\times\!10^6 = 5.99\!\times\!10^{-14}$ W $= -132.2$ dBm.
(a) $G/T = G_r - 10\log T_{sys} = 45 - 10\log 120 = 45 - 20.79 = 24.2$ dB/K.
(b) Link budget in dB:
$C/N_0 = $ EIRP $-$ FSPL $+ G/T - 10\log k$
$= 50 - 205 + 24.2 - (-228.6) = 97.8$ dB-Hz.
(c) For e.g. a 36 MHz transponder: $C/N = C/N_0 - 10\log B = 97.8 - 75.6 = 22.2$ dB — comfortably enough for 16-APSK / DVB-S2.
$T_{TL} = (L_T - 1)T_p$ with $L_T = 10^{0.05} = 1.122$ → $T_{TL} = 0.122 \times 290 = 35.4$ K.
$T_{SYS} = T_a + T_{TL} + L_T \cdot T_{REC} = 200 + 35.4 + 1.122 \times 180 = 200 + 35.4 + 202 = 437.4$ K.
$N_{SYS} = kT_{SYS}B = 1.38\!\times\!10^{-23}\times 437.4 \times 2\!\times\!10^7 = 1.21\!\times\!10^{-13}$ W $= -99.2$ dBm.
$\text{SNR}_{out} = S_i - N_{SYS} = -80 - (-99.2) = 19.2$ dB.
Substitute $v_i = A\cos\omega t$:
Method
First compute the output predicted by the small-signal straight-line gain, then subtract the 1 dB compression error.
Small-signal extrapolated output at $P_{in}=-10$ dBm is $P_{out,ideal}=P_{in}+G=-10+20=10$ dBm.
At the 1 dB compression point, the actual output is 1 dB lower than this extrapolated value: $\text{OP1dB}=10-1=9$ dBm.
Set $\Delta = f_2 - f_1 = 1$ MHz.
2nd-order: $2f_1 \approx 2$ GHz, $2f_2 \approx 2$ GHz, $f_1 + f_2 \approx 2$ GHz, $f_2 - f_1 = 1$ MHz — all out-of-band.
3rd-order: $3f_1, 3f_2 \approx 3$ GHz (out); $2f_1 \pm f_2$ → $0.999, 3.001$ GHz; $2f_2 \pm f_1$ → $1.002, 3.002$ GHz.
$\Delta P = P_{fund} - P_{IM3} = -10 - (-70) = 60$ dB.
OIP3 $= P_{fund} + \Delta P/2 = -10 + 30 = 20$ dBm.
Slope rule: on a $\log P_{out}$ vs $\log P_{in}$ plot, fund grows 1 dB / dB; IM3 grows 3 dB / dB. They intersect at IP3. At $\Delta$ dB backoff from IP3, fund is $\Delta$ dB below IP3 and IM3 is $3\Delta$ dB below — gap $= 2\Delta$.
Here the measured gap is 60 dB, so the fundamental is $60/2=30$ dB below the intercept. Since the measured fundamental is $-10$ dBm, the intercept is $-10+30=20$ dBm.
(a) $P_{out,fund} = P_{in} + G = -30 + 20 = -10$ dBm/tone.
(b) $P_{IM3} = 3 P_{out} - 2\,\text{OIP3} = 3(-10) - 2(30) = -30 - 60 = -90$ dBm.
Backoff check: the fundamental is $30-(-10)=40$ dB below OIP3. IM3 is twice that gap below the fundamental, so IM3 should be $2\cdot40=80$ dBc. $-10-80=-90$ dBm, matching the formula.
(a) OP1dB $= 22$ → OIP3 $\approx 22 + 10.6 = 32.6$ dBm.
(b) OIP3 $= 35$ → OP1dB $\approx 35 - 10.6 = 24.4$ dBm.
(c) Breakdown cases:
(a) $DR_l = $ OP1dB $- N_o = 20 - (-90) = 110$ dB.
(b) $DR_f = (2/3)($ OIP3 $- N_o) = (2/3)(120) = 80$ dB.
(c) $DR_l - DR_f = 30$ dB. The "$\approx 21.3$ dB" rule from §3.6 holds when OIP3 $-$ OP1dB $= 10.6$ dB and $N_o$ tracks the canonical $kT_0 B F$ floor; here OIP3 $-$ OP1dB $= 10$ dB and $N_o$ was specified independently, so the gap differs slightly from the rule.
Convert: OIP3$_{,1} = 20$ dBm $= 100$ mW, $G_1 = 10$ dB $=10$, $G_2 = 20$ dB $=100$, OIP3$_{,2} = 25$ dBm $= 316$ mW.
$\dfrac{1}{\text{OIP3}} = \dfrac{1}{G_2 \cdot \text{OIP3}_{,1}} + \dfrac{1}{\text{OIP3}_{,2}} = \dfrac{1}{100\cdot 100} + \dfrac{1}{316} = 10^{-4} + 3.16\!\times\!10^{-3} = 3.26\!\times\!10^{-3}$.
OIP3 $= 307$ mW $= 24.9$ dBm. Stage 2 dominates ($1/316$ term is 31× larger).
Linear: $G_1=31.62$, $G_2=0.200$, $G_3=316.2$. OIP3$_{,1}=31.6$ mW, OIP3$_{,2}=63.1$ mW, OIP3$_{,3}=631$ mW.
$\dfrac{1}{\text{OIP3}} = \dfrac{1}{\text{OIP3}_{,1} G_2 G_3} + \dfrac{1}{\text{OIP3}_{,2} G_3} + \dfrac{1}{\text{OIP3}_{,3}}$
$= \dfrac{1}{31.6 \cdot 0.2 \cdot 316.2} + \dfrac{1}{63.1 \cdot 316.2} + \dfrac{1}{631}$
$= \dfrac{1}{1998} + \dfrac{1}{19952} + \dfrac{1}{631} = 5.0\!\times\!10^{-4} + 5.0\!\times\!10^{-5} + 1.585\!\times\!10^{-3} = 2.14\!\times\!10^{-3}$.
OIP3 $= 467$ mW $= 26.7$ dBm.
IIP3 $= $ OIP3 $- G = 24 - 12 = 12$ dBm.
Input-referred: $P_{IM3,in} = 3 P_{in} - 2\,\text{IIP3} = 3(-20) - 2(12) = -60 - 24 = -84$ dBm.
(Sanity-check at output: $P_{out,fund} = -8$ dBm, $P_{out,IM3} = 3(-8) - 2(24) = -72$ dBm = $-84 + G$ ✓.)
ACPR = $10\log\dfrac{P_{adj\,channel}}{P_{main\,channel}}$ for a digitally modulated test signal — power that "leaks" into a defined adjacent channel BW.
For weakly-nonlinear amps the ACPR floor is set by 3rd-order products generated by all the in-band tones convolving with each other. Same physics as 2-tone IMR but with a continuum of tones.
Empirically, the ACPR test "sees" the integrated 3rd-order power vs 2-tone's peak product, so ACPR specs typically demand $\sim 3$–$4$ dB more OIP3 than the equivalent 2-tone IMR target.
(a) Input noise floor: $N_i = kT_0 B F = -174 + 60 + 4 = -110$ dBm. Output: $N_o = N_i + G = -110 + 60 = -50$ dBm.
(b) IIP3 $= $ OIP3 $- G = 30 - 60 = -30$ dBm.
(c) Input-referred SFDR $= (2/3)($ IIP3 $- N_i) = (2/3)(-30 - (-110)) = (2/3)(80) = 53.3$ dB.
Interferer scenario: $-30$ dBm interferer is right at IIP3 — IM3 generated will be $3(-30) - 2(-30) = -30$ dBm at input… that's the IP3 itself, completely overwhelming a $-90$ dBm desired signal. The receiver fails: gap $= -30 - (-90) = 60$ dB > SFDR (53.3 dB).
Use $\cos^2 = \tfrac{1}{2}(1+\cos 2\omega t)$ and $\cos^3 = \tfrac{3}{4}\cos\omega t + \tfrac{1}{4}\cos 3\omega t$. With $V_0 = 0.4$ V:
(a) DC: $\tfrac{1}{2}a_2 V_0^2 = 0.5 \times 0.3 \times 0.16 = 0.024$ V.
(b) Fundamental amplitude: $a_1 V_0 + \tfrac{3}{4}a_3 V_0^3 = 5 \times 0.4 + 0.75 \times (-0.05) \times 0.064 = 2.0 - 0.0024 = 1.998$ V.
(c) 2nd harmonic: $\tfrac{1}{2}a_2 V_0^2 = 0.024$ V at $2\omega_0$.
(d) 3rd harmonic: $\tfrac{1}{4}a_3 V_0^3 = 0.25 \times (-0.05) \times 0.064 = -8\!\times\!10^{-4}$ V (i.e. $0.8$ mV at $3\omega_0$).
(e) Effective gain at $\omega_0$: $G_v = a_1 + \tfrac{3}{4}a_3 V_0^2 = 5 - 0.006 = 4.994$ ≈ 14 dB. Compression $\approx 0.01$ dB.
(a) Linear region: $P_{out} = P_{in} + G = P_{in} + 18$. As $P_{in}$ rises, gain compresses; at $P_{1dB,out} = 10$ dBm, gain has dropped to 17 dB.
(b) $P_{1dB,in} = P_{1dB,out} - (G - 1) = 10 - 17 = -7$ dBm. At $P_{in} = -8$ dBm, the device is just below 1 dB compression — actual $G \approx 17.6$ dB → $P_{out} \approx -8 + 17.6 = 9.6$ dBm.
(c) $P_{1dB,in} = -7$ dBm.
$f_1 = 2400$, $f_2 = 2410$ MHz. Passband $2355\text{–}2455$ MHz.
IM2: $f_2 - f_1 = 10$ MHz (well below passband); $f_1 + f_2 = 4810$ MHz (well above); $2 f_1 = 4800$, $2 f_2 = 4820$ MHz (above). None fall in band.
IM3: $2f_1 - f_2 = 2390$ MHz, $2f_2 - f_1 = 2420$ MHz — both in-band (next to fundamentals). $3f_1 = 7200$ MHz, $3f_2 = 7230$ MHz, $2f_1 + f_2 = 7210$ MHz, $f_1 + 2f_2 = 7220$ MHz — far above passband.
The two in-band products $2f_1 - f_2$ and $2f_2 - f_1$ both come from the cubic $a_3$ term.
Slope-1 fundamental, slope-3 IM3 lines on a log-log $P_{out}$ vs $P_{in}$ chart.
Gap at $P_{in} = -10$ dBm: $\Delta = P_{out,fund} - P_{IM3} = 0 - (-50) = 50$ dB.
$\text{IIP}_3 = P_{in} + \Delta/2 = -10 + 25 = 15$ dBm.
$\text{OIP}_3 = \text{IIP}_3 + G = 15 + 10 = 25$ dBm.
IM3 ratio = 50 dBc — adequate for moderate-linearity systems but marginal for cellular base stations (which require > 70 dBc typically).
(a) $OP_{1dB} \approx \text{OIP}_3 - 10.6 = 28 - 10.6 = 17.4$ dBm.
(b) $IP_{1dB,in} = OP_{1dB} - G_{1dB} \approx OP_{1dB} - (G - 1) = 17.4 - 24 = -6.6$ dBm.
(c) $\text{IIP}_3 = \text{OIP}_3 - G = 28 - 25 = 3$ dBm.
Limitation: the 10.6 dB constant comes from the strict cubic Taylor model. Real devices with significant 5th-order content, AM-PM distortion, or memory effects can deviate by $\pm 2$ dB or more — always verify with two-tone measurement when accuracy matters.
(a) $N_o = -174 + 10\log B + F + G = -174 + 60 + 4 + 30 = -80$ dBm.
(b) $DR_l = OP_{1dB} - N_o = 10 - (-80) = 90$ dB.
(c) $DR_f = (2/3)(\text{OIP}_3 - N_o) = (2/3)(20 - (-80)) = (2/3)(100) = 66.7$ dB.
(d) $DR_l - DR_f = 90 - 66.7 = 23.3$ dB ≈ 21 dB ✓ (the $\sim 21.3$ dB rule).
Convert: $G_1 = 100$, $L = 4$ (so $G_2 = 0.25$), OIP3$_1 = 31.6$ mW, OIP3$_2 = 100$ mW.
$1/\text{OIP}_3 = 1/(G_2 \cdot \text{OIP}_3^{(1)}) + 1/\text{OIP}_3^{(2)} = 1/(0.25\times 31.6) + 1/100 = 1/7.9 + 1/100 = 0.1266 + 0.01 = 0.1366$.
OIP3$_{cas} = 7.32$ mW $= 8.65$ dBm.
Mixer dominates linearity ($1/(G_2\cdot \text{OIP}_3^{(1)}) = 1/7.9$ is 12.7× larger than $1/\text{OIP}_3^{(2)}$).
(a) PIM3 / single carrier: $-125 - 43 = -168$ dBc — a typical good base-station spec ($< -150$ dBc).
(b) PIM3 vs noise floor: $-125 - (-110) = -15$ dB → PIM3 is $15$ dB below the receiver noise floor. Just below the noise — would corrupt sensitivity if it rose any further.
(c) PIM is generated by passive nonlinearities (bad connectors, oxidation, ferromagnetic parts). It only matters when powers are large — i.e. on the transmit side. The receiver's small signal levels can't generate appreciable PIM, so receivers don't worry about it. But the same antenna often sees both Tx and Rx signals (FDD), so PIM generated by the Tx can fall into the Rx band.
| Symbol | Name | Units | Notes |
|---|---|---|---|
| $S(f)$, $S(\omega)$ | Power spectral density (PSD) | W/Hz | Two-sided. Total power $P = \int S(f)\,df$ |
| $R(\tau)$ | Autocorrelation function | W (or V²) | $R(\tau) = \mathcal{F}^{-1}\{S(f)\}$. $R(0)$ = total power |
| $n_0$ | Noise spectral density (one-sided) | W/Hz | $n_0 = kT$. Two-sided is $n_0/2$ |
| $\delta(\tau)$ | Dirac delta function | 1/s | White noise: $R(\tau) = (n_0/2)\delta(\tau)$ (uncorrelated) |
| $\sigma$, $\sigma^2$ | Noise standard deviation / variance | V / V² | Gaussian PDF: $p(v) = \tfrac{1}{\sqrt{2\pi}\sigma}e^{-v^2/2\sigma^2}$ |
| $V$ or $v_0$ | Peak signal voltage | V | PSK signal $s(t)=\pm V\cos\omega_c t$ |
| $T$ | Bit/symbol period | s | $T = 1/R_b$ for binary signaling |
| $E_b$ | Energy per bit | J | $E_b = V^2 T/2$ for sinusoidal PSK |
| $P_e$ | Bit error probability (BER) | unitless | Typical target $10^{-5}$ to $10^{-9}$ |
| $\mathrm{erfc}(x)$ | Complementary error function | unitless | $\mathrm{erfc}(x) = 1 - \mathrm{erf}(x)$, Gaussian-tail probability |
| $\omega_c$, $f_c$ | Carrier (angular) frequency | rad/s, Hz | $\omega_c = 2\pi f_c$ |
| $M$ | Modulation order | integer | M-QAM, M-PSK; bits/symbol $= \log_2 M$ |
| Symbol | Name | Units | Notes |
|---|---|---|---|
| $k$ | Boltzmann constant | $1.38\times10^{-23}$ J/K | $10\log(k) = -228.6$ dBW/Hz/K |
| $T_0$ | Reference temperature | $290$ K | Used in noise-figure definition; $kT_0 = -174$ dBm/Hz |
| $B$ | Noise bandwidth | Hz | IF/channel bandwidth that defines noise power |
| $T_e$ | Equivalent noise temperature | K | Of receiver/component; $T_e = (F-1)T_0$ |
| $F$ | Noise figure (factor) | linear / dB | $F = \text{SNR}_{in}/\text{SNR}_{out}$ at $T_0=290$ K |
| $G$ | Available power gain | linear / dB | Antenna gain in linear or dBi |
| $L$ | Loss factor | $\ge 1$ / dB | Passive lossy line at $T_p$: $T_e=(L-1)T_p$, $F=L$ when $T_p=T_0$ |
| $T_b$ | Brightness (background) temperature | K | Beam-weighted: cold sky 3–50 K, ground ~290 K |
| $T_p$ | Physical temperature | K | Of antenna structure / lossy line |
| $T_a$ / $T_A$ | Antenna noise temperature | K | $T_a = \eta_{rad}T_b + (1-\eta_{rad})T_p$ |
| $\eta_{rad}$ | Radiation efficiency | $0$–$1$ | Fraction radiated; $1-\eta_{rad}$ dissipated |
| $T_{SYS}$ | System noise temperature | K | $T_{SYS} = T_a + T_e$ (referred to antenna) |
| EIRP | Equivalent Isotropic Radiated Power | W / dBW | $\text{EIRP} = G_t P_t$ |
| FSPL | Free-space path loss | dB | $20\log_{10}(4\pi R/\lambda)$ |
| $G/T$ | Antenna figure of merit | dB/K | $G_r(\text{dBi}) - 10\log_{10}(T_a + T_e)$ |
| $E_b/n_0$ | Energy-per-bit / noise PSD | dB | Sets BER independent of modulation |
| $R_b$ | Bit rate | bps | $\text{CNR} = (E_b/n_0)(R_b/B)$ |
| P1dB | 1 dB compression point | dBm | Input or output (IP1dB / OP1dB) |
| IP3 / OIP3 | 3rd-order intercept point | dBm | Input/output; OIP3 ≈ OP1dB + 10.6 dB |
| $DR_l$, $DR_f$ | Linear DR / Spurious-free DR | dB | $DR_l - DR_f \approx 21.3$ dB |
$P = \displaystyle\int_{-\infty}^{\infty} S(f)\,df$
Time avg = area under PSD
$S(f) \;\longleftrightarrow\; R(\tau)$ (Fourier pair)
$R(0)$ = total power
$S(f) = n_0/2$ (two-sided)
$R(\tau) = (n_0/2)\,\delta(\tau)$
$n_0 = kT$
$P_n = n_0 B = kTB$
Integrated over physical $B$
$p(v) = \dfrac{1}{\sqrt{2\pi}\,\sigma}\,e^{-v^2/2\sigma^2}$
Tail: $P(v>V)=\tfrac{1}{2}\mathrm{erfc}(V/\sqrt{2}\sigma)$
$E_b = \dfrac{V^2 T}{2}$ (PSK, sinusoidal)
$E_b = P_s \cdot T_b$
Output signal: $VT$
$\sigma_{out}^2 = n_0/T$
$\Rightarrow E_b/n_0$ ratio
$P_e = \tfrac{1}{2}\mathrm{erfc}\!\left(\sqrt{E_b/n_0}\right)$
$10^{-5}$ → $E_b/n_0 = 9.6$ dB
$\eta_s = R_b/B$ (bps/Hz)
BPSK: 1, QPSK: 2, 16-QAM: 4, 64-QAM: 6
$\dfrac{S}{N} = \dfrac{E_b}{n_0}\cdot\dfrac{R_b}{B}$
Bridges symbol-rate and bandwidth views
Coherent PSK: 9.6 dB
Coherent FSK: 12.6 dB; Noncoh: 13.4 dB
Coherent ASK: 15.6 dB; Noncoh: 16.4 dB
16-QAM: $E_b/n_0 \approx 14.4$ dB
64-QAM: $E_b/n_0 \approx 22.5$ dB
SSB: SNR unchanged after demod
DSB-SC (coherent): +3 dB SNR gain
$\eta = \dfrac{m^2}{2+m^2}$, $m \le 1$
Max 33% at $m=1$ (carrier wastes power)
$P_n = kTB$
$v_n = \sqrt{4kTBR}$
$kT_0 = -174$ dBm/Hz
$F = 1 + T_e/T_0$
$T_e = (F-1)T_0$
$T_0 = 290$ K
$F = F_1 + \dfrac{F_2-1}{G_1} + \dfrac{F_3-1}{G_1G_2} + \cdots$
1st stage dominates → low-NF, high-gain LNA first
$T_e = T_1 + \dfrac{T_2}{G_1} + \dfrac{T_3}{G_1G_2} + \cdots$
$T_e = (L-1)T_p$
$F = L$ when $T_p = T_0$
$F$(dB) $=$ IL(dB)
$T_a = \eta_{rad}T_b + (1-\eta_{rad})T_p$
Lossless: $T_a = T_b$
$T_{SYS} = T_a + T_{TL} + L_T\,T_{REC}$
$T_{TL} = (L_T-1)T_p$
$P_r = P_t G_t G_r\!\left(\dfrac{\lambda}{4\pi R}\right)^{\!2}$
EIRP $= G_t P_t$
FSPL $= 20\log_{10}\!\left(\dfrac{4\pi R}{\lambda}\right)$
$= 32.4 + 20\log f_{\text{MHz}} + 20\log R_{\text{km}}$
$G/T$ (dB/K) $= G_r$(dBi) $- 10\log_{10}(T_a + T_e)$
CNR $= \text{EIRP} + G/T - \text{FSPL} - 10\log(kB)$
$10\log(kB) = -228.6 + 10\log B$
$S_{i,min} = kB[T_a + (F-1)T_0]\,(S_o/N_o)_{min}$
$S/N = (E_b/n_0)(R_b/B)$
BPSK/QPSK: $E_b/n_0 = 9.6$ dB → BER $10^{-5}$
$P_e = \tfrac{1}{2}\mathrm{erfc}\!\left(\sqrt{E_b/n_0}\right)$
OIP3 $\approx$ OP1dB $+ 10.6$ dB
IM3 slope = 3:1, fundamental = 1:1
$\dfrac{1}{\text{OIP}_3} = \dfrac{1}{G_2\,\text{OIP}_{3,1}} + \dfrac{1}{\text{OIP}_{3,2}} + \cdots$
Last stage dominates (opposite of noise)
$DR_l = $ P1dB $- N_o$
$DR_f = \tfrac{2}{3}(\text{OIP}_3 - N_o)$
$DR_l - DR_f \approx 21.3$ dB
$T_e = \dfrac{T_h - YT_c}{Y - 1}$
$Y = N_h/N_c$
Part 1 — Random Processes, PSD & Modulation
Part 2 — Noise Figure, Antenna & Distortion