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Dsp Audio Rf And Images

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Use when working in a specific signal domain: audio and speech (psychoacoustics, coding and codecs, the real-time voice pipeline with echo cancellation, noise suppression and latency budgets, music and analysis), RF and communications (modulation, software-defined radio, synchronization, equalization), and images and video treated as two-dimensional signals.

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  • Added September 19, 2026
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SKILL.md
---
name: dsp-audio-rf-and-images
description: "Use when working in a specific signal domain: audio and speech (psychoacoustics, coding and codecs, the real-time voice pipeline with echo cancellation, noise suppression and latency budgets, music and analysis), RF and communications (modulation, software-defined radio, synchronization, equalization), and images and video treated as two-dimensional signals."
---

# Signal Processing: Audio and Speech, RF and Communications, and Images and Video

> **Part 3 of 5** of the *Signal Processing* reference (plugin `signal-processing`), covering §8–§10. Sibling skills: `dsp-sampling-frequency-domain-and-filters` (§0–§3), `dsp-convolution-multirate-and-spectral-analysis` (§4–§7), `dsp-implementation-tools-and-testing` (§11–§14), `dsp-reference` (§15–§20). Section numbers are shared across the set; a reference written as §N → `skill` points into that sibling skill.
>
> **Currency:** Verified August 2026. See §17 → `dsp-reference` for the currency snapshot and what goes stale first.

> **How to read this.** Reference for engineers who need DSP working, not a course. Three
> markers:
> - **[DURABLE]** — sampling theory, transforms, filters, estimation. **Nyquist is 1928;
>   Shannon 1949; the Cooley–Tukey FFT 1965. This material does not expire.**
> - **[VERSIONED]** — codecs, libraries, hardware, the ML layer.
> - **[CONTESTED]** — genuine disagreement.
>
> **⚠️ GOTCHA** boxes mark where the code runs, the plot looks fine, and the answer is
> wrong.
>
> **The three framings that organize everything below:**
> 1. **Everything is a trade between time and frequency resolution, and you cannot cheat
>    it.** The uncertainty principle is not a metaphor here — **a short window gives you
>    good time resolution and poor frequency resolution, and no amount of cleverness
>    escapes that** (§2.4 → `dsp-sampling-frequency-domain-and-filters`). Most DSP design is choosing where on that curve to sit.
> 2. **⚠️ Aliasing is irreversible.** Once you've sampled too slowly, the information is
>    gone — no filter, no algorithm, and no neural network recovers it. **This is the one
>    mistake in this document that cannot be fixed downstream** (§1 → `dsp-sampling-frequency-domain-and-filters`).
> 3. **The field is mid-convergence with machine learning, and the interesting result is
>    that hybrids are winning** — not pure DSP, and not pure end-to-end learning.
>    ⚠️ **The pattern that keeps recurring: use DSP for the structure you know, and learn
>    the parameters you can't hand-tune** (§8, §16.1 → `dsp-reference`).

---

## §8. Audio and Speech

### 8.1 The domain facts

**[DURABLE]** Human hearing spans roughly **20 Hz – 20 kHz** (⚠️ **and the top end
declines substantially with age**), with **~120 dB of dynamic range**, **logarithmic**
in both frequency and amplitude. **Critical bands / Bark / ERB scales** model the ear's
frequency resolution, and **masking** — a loud tone hiding a nearby quiet one, in frequency
and in time — is the entire basis of perceptual audio coding.

⚠️ **Latency budgets are what constrain audio engineering**: musicians notice round-trip
latency above roughly **10 ms**; conversational latency becomes awkward above ~**150–200 ms**
one-way. **These numbers determine your block size, and block size determines everything
else.**

### 8.2 Coding

**Perceptual codecs** (MP3, AAC, Vorbis, Opus) throw away what masking says you can't
hear: **MDCT** transform, psychoacoustic model, quantize, entropy-code.
**Speech codecs** (⚠️ **model the vocal tract rather than the waveform**): LPC, CELP,
AMR, and Opus's SILK layer.

**[VERSIONED] Opus is the one to know** — RFC 6716, hybrid SILK + CELT, **mandatory in
WebRTC**, and covers everything from speech at low bitrate to full-band music.

> **⚠️ GOTCHA — and the most interesting story in DSP right now: classical codecs are
> absorbing neural components rather than being replaced by them.**
>
> **Opus 1.5 (March 2024) put machine learning inside the codec for the first time**, all
> at the decoder — meaning **a service can adopt it on the playback side alone** and
> improve calls for users on bad networks without changing the sender.
> - **Deep PLC** — a neural network reconstructs lost packets, in place of the classical
>   repeat-and-fade. Recommended for occasional loss.
> - **DRED (Deep REDundancy)** — for **burst** loss. It uses a **rate-distortion-optimized
>   VAE** to compress acoustic features so aggressively that **up to one second of
>   redundancy fits in ~12–32 kb/s of overhead**, carried in each packet's padding.
>   ⚠️ **It transmits 20 acoustic features — 18 Bark-frequency cepstral coefficients plus
>   pitch and voicing — and resynthesizes speech from them.** Note the consequence: **the
>   features contain no phase information, so the recovered waveform deviates
>   significantly from the original while sounding similar.**
> - **LACE/NoLACE** — a DNN post-filter that denoises the decoder's output.
>
> **Opus 1.6 (Dec 2025)** added **bandwidth extension (BWE)**, which combined with DRED
> reaches **fullband quality**, and with NoLACE enables **good fullband speech as low as
> 9 kb/s.**
>
> ⚠️ **Two caveats worth carrying.** **DRED's format was still being finalised in IETF as
> of 2026 — treat it as advanced rather than settled.** And **any DNN in a codec
> significantly increases real-time compute**, which is the actual adoption barrier —
> along with model size, which Opus's own developers name as a main obstacle.

**Neural codecs proper** — SoundStream, EnCodec, and the discrete-token codecs underpinning
generative audio — are a genuinely different lineage: much lower bitrates, much higher
compute, and **tokens that double as inputs to generative models.**

### 8.3 The real-time voice pipeline

**[DURABLE] The order matters and is roughly fixed**: **AEC → noise suppression →
dereverberation → AGC → codec.**

**Acoustic echo cancellation** — ⚠️ **adaptive filter (§6 → `dsp-convolution-multirate-and-spectral-analysis`) to model the echo path, plus a
nonlinear residual echo suppressor, plus a double-talk detector.** Interestingly, **most
state-of-the-art AEC is still classical DSP or hybrid DSP-ML**, with deep learning
typically replacing only the nonlinear residual suppressor.

**Noise suppression — and this is where the convergence story is clearest.** The classical
line runs **spectral subtraction (Boll)** → ⚠️ **which produced the notorious "musical
noise" artifact** → **Wiener filtering** (needs a priori statistics) → **MMSE log-spectral
amplitude (Ephraim–Malah, 1985)**, still a respectable baseline.

**RNNoise (Valin, 2018) is the landmark hybrid**: a small recurrent network estimates
**ideal critical-band gains** while **a traditional pitch filter attenuates noise between
harmonics** — ⚠️ **significantly better than a classical MMSE estimator while staying
real-time at 48 kHz on a low-power CPU.** **DeepFilterNet** and successors extended this by
**predicting linear filters rather than estimating clean speech directly**, which is
computationally far cheaper and edge-deployable.

**⚠️ The honest current comparison**: traditional approaches are **fast but conservative**;
deep learning can be **more aggressive while preserving speech quality, at higher
computational cost.** In hearing-aid studies, deep and classical methods perform
**similarly in diffuse noise**, with the deep binaural approach winning **in the presence
of spatial interferers.** **The pattern throughout: hybrids, not replacements.**

**Beamforming** for multi-mic arrays — delay-and-sum, MVDR, GSC — ⚠️ **spatial filtering
that is genuinely complementary to spectral methods**, and increasingly paired with neural
mask estimation.

### 8.4 Music and analysis
**Pitch detection** (autocorrelation, YIN, CREPE), **onset detection**, **beat tracking**,
**source separation** (⚠️ **Demucs, Spleeter and the neural separators substantially beat
classical NMF-based methods** — one of the cleanest ML wins in audio), **MFCCs** (⚠️ **the
classical speech feature; largely superseded by learned representations and mel
spectrograms in modern ML pipelines, but still everywhere in legacy code**), **time-scale
and pitch modification** (phase vocoder, PSOLA, WSOLA).

---

## §9. RF and Communications

**Modulation**: analogue (AM/FM/PM), digital (**ASK/FSK/PSK/QAM**), and **OFDM** —
⚠️ **which converts a frequency-selective channel into many flat sub-channels and is why
it's in Wi-Fi, LTE, 5G, and DVB.** **Spread spectrum** (DSSS, FHSS) for interference
resistance and multiple access.

**The receive chain**: down-conversion to **I/Q baseband** (⚠️ **complex-valued
representation is the whole language of RF DSP — get comfortable with it**),
**matched filtering** (§4 → `dsp-convolution-multirate-and-spectral-analysis`), **timing and carrier recovery**, **equalization** (§6 → `dsp-convolution-multirate-and-spectral-analysis`), and
**demodulation**.

**Error correction**: Hamming and Reed–Solomon (⚠️ **still the standard for burst errors in
storage and broadcast**), convolutional codes with Viterbi decoding, **Turbo**, **LDPC**
(⚠️ **near-Shannon-limit, and in 5G, Wi-Fi 6 and DVB-S2**), and **polar codes** in 5G
control channels.

**⚠️ The physical realities that dominate**: multipath and fading, Doppler, the noise
figure of your front end, **and the fact that Shannon capacity is a hard ceiling** —
C = B log₂(1 + SNR). **No modulation scheme beats it.**

**SDR** is where software engineers actually meet this: **GNU Radio** (flowgraph-based),
**RTL-SDR** (⚠️ **~$30 and genuinely capable for receive-only experimentation**),
**HackRF**, **USRP**, **LimeSDR**, and **SoapySDR** as the hardware abstraction.
⚠️ **Transmitting requires a licence in most jurisdictions and most bands — receiving
generally doesn't.**

---

## §10. Images and Video

**[DURABLE] 2D signal processing, and the concepts transfer directly.**
**2D convolution** for blur, sharpen, and edge detection (Sobel, Laplacian);
**separable kernels** (⚠️ **a 2D Gaussian is two 1D passes — O(n) instead of O(n²) per
pixel, and this is why Gaussian blur is cheap**); **2D FFT** for frequency-domain
filtering; **the DCT** as JPEG's core.

**⚠️ Aliasing appears here as moiré patterns** — and image downsampling without
pre-filtering is exactly §1.1 → `dsp-sampling-frequency-domain-and-filters`'s error in two dimensions. **Anti-aliasing in rendering is
the same problem approached from the synthesis side.**

**Also**: **image pyramids** (Gaussian, Laplacian) for multi-scale processing;
**bilateral and non-local means** filters (⚠️ **edge-preserving — linear filters can't do
this**); **optical flow** for motion; **video codecs** as motion-compensated transform
coding (H.264/AVC, H.265/HEVC, AV1, and H.266/VVC).

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