Block Diagram of a Digital Communication System

Block Diagram Of Digital Communication System — circuit diagram showing component connections+InputAStage 1+-Stage 2ARDUINOUNOProcessorOutputBlock Diagram Of Digital Communication System
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A block diagram of a digital communication system shows the sequential processing stages — source, source encoder, channel encoder, modulator, channel, demodulator, channel decoder, source decoder, and destination — that convert information into transmitted signals and back.

A digital communication system converts information from a source into a digital bit stream, processes that stream to make it suitable for transmission, sends it through a physical channel, and reconstructs the original information at the receiver. Every stage addresses one specific problem in the chain.

Source and Source Encoder: the source generates information — speech, video, data, or text. The source encoder compresses the information by removing statistical redundancy. Examples include MP3/AAC for audio (psychoacoustic coding), H.264/H.265 for video (transform coding and motion compensation), and Huffman or arithmetic coding for lossless data compression. The output is a compact digital bit stream.

Channel Encoder: redundancy is deliberately added back in a controlled mathematical way to protect the data against transmission errors. Forward error correction (FEC) codes such as Hamming codes, Reed–Solomon codes, convolutional codes, turbo codes, and LDPC (Low-Density Parity-Check) codes allow the receiver to detect and correct bit errors without retransmission. This is distinct from source coding — source coding removes redundancy; channel coding adds structured redundancy.

Modulator: the encoded bit stream is mapped onto a physical waveform suitable for the transmission channel. Modulation schemes include ASK (amplitude-shift keying), FSK (frequency-shift keying), PSK (phase-shift keying), and QAM (quadrature amplitude modulation). Higher-order schemes (e.g., 256-QAM) carry more bits per symbol but require a better signal-to-noise ratio (SNR). Spread-spectrum techniques (DSSS, FHSS) are used for resilience or multiple access.

Channel: the physical medium through which the signal travels — twisted pair, coaxial cable, optical fibre, radio spectrum, or acoustic channel. The channel adds noise (typically modelled as additive white Gaussian noise, AWGN), introduces multipath fading in wireless links, limits bandwidth, and may introduce non-linear distortion.

Demodulator, Channel Decoder, Source Decoder: at the receiver, the demodulator recovers the digital bit stream from the received waveform. The channel decoder applies the inverse of the FEC code, correcting errors within the code's capability. The source decoder reconstructs the original information from the compressed representation. The destination receives the reconstructed output — audio, video, or data.

How to wire block diagram of digital communication system

  1. Define the information source and its statistical properties Identify the type and rate of information to be transmitted: discrete symbols (text, binary data), continuous analogue signals requiring analogue-to-digital conversion (voice, audio), or inherently digital data (computer files, sensor readings). Characterise the source entropy — the minimum bits per symbol required, which governs source coding efficiency.
  2. Design the source encoder Select a source coding algorithm appropriate to the data type and required fidelity. For lossless compression of discrete data, use entropy coding (Huffman, arithmetic coding, LZ77/DEFLATE). For lossy compression of audio, use perceptual coding (MP3, AAC, Opus). For video, use predictive and transform coding (H.264, H.265, AV1). The target output is a compact bit stream at the minimum necessary rate.
  3. Select and design the channel code Determine the expected channel bit error rate (BER) from channel characteristics. Choose a FEC code with sufficient error-correcting capability: simple Hamming codes for low-error links; Reed–Solomon codes for burst-error channels (optical storage, satellite); convolutional codes for mobile channels; turbo or LDPC codes for near-Shannon-limit performance in mobile broadband. The code rate (k/n, where k data bits become n coded bits) determines the overhead imposed.
  4. Choose the modulation scheme Match the modulation scheme to the channel bandwidth and required SNR. BPSK and QPSK are robust at low SNR (satellite, deep-space). Higher-order QAM (16-QAM, 64-QAM, 256-QAM, 1024-QAM) achieves high spectral efficiency at good SNR — used in Wi-Fi and 4G/5G. OFDM (orthogonal frequency-division multiplexing) divides the bandwidth into many parallel subcarriers, combating multipath fading and enabling simple equalisation — it underlies LTE, 5G NR, Wi-Fi, and DVB.
  5. Model and characterise the channel Determine the channel model: AWGN for wired or line-of-sight links; Rayleigh fading for mobile radio without a line-of-sight path; Rician fading when a dominant line-of-sight exists; frequency-selective fading for wideband channels. Measure or estimate available bandwidth, expected SNR, and whether the channel is time-varying. These parameters directly constrain achievable capacity via the Shannon–Hartley theorem.
  6. Design the receiver chain The demodulator must match the modulator: it samples the received waveform at the symbol rate, determines the most likely transmitted symbol (using maximum likelihood detection or equivalent), and passes soft or hard decisions to the channel decoder. Soft decisions (log-likelihood ratios) significantly improve turbo and LDPC decoder performance. The channel decoder reverses the FEC to recover the source-coded bit stream.
  7. Evaluate system performance Simulate the complete chain using a BER versus Eb/N0 curve. Compare against the theoretical Shannon limit. Verify the source decoder output quality (for lossy coding, use PSNR for images/video or PESQ/POLQA for speech). Iterate on the modulation order, code rate, and frame structure until the performance targets are met within the bandwidth and power constraints.

Specifications

Shannon channel capacity formulaC = B × log₂(1 + SNR) bits per second
BPSK spectral efficiency1 bit/s/Hz
QPSK spectral efficiency2 bits/s/Hz
64-QAM spectral efficiency6 bits/s/Hz
256-QAM spectral efficiency8 bits/s/Hz
Nyquist minimum sampling rate2 × maximum signal frequency (fs ≥ 2B)
Typical LDPC/turbo code performanceWithin 0.5–1.5 dB of Shannon limit at practical code lengths
Typical BER target for voice/data services10⁻³ for compressed voice; 10⁻⁶ to 10⁻⁹ for data services (application dependent)

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Tools needed

Common mistakes

Troubleshooting

Bit error rate much higher than predicted by simulation
Cause: Channel model mismatch (actual channel has more fading or interference than modelled), synchronisation failure, ADC saturation/clipping, or implementation loss in the demodulator Fix: Measure actual channel SNR with a calibrated receiver. Check ADC input level — clip events dramatically raise BER. Verify carrier frequency offset and timing synchronisation loops are converging. Compare measured BER curve against theoretical AWGN and fading models to identify whether the gap suggests an SNR shortfall or an implementation defect.
Decoder fails to correct errors; BER does not improve after decoding
Cause: Number of errors per codeword exceeds the decoder's error-correcting capacity, or decoder is receiving hard decisions instead of soft decisions (for turbo/LDPC) Fix: Check the channel BER entering the decoder against the code's design error correction capability. For turbo or LDPC decoders, verify that soft-decision (LLR) values are being passed correctly rather than hard-quantised bits. Increase the channel SNR or switch to a stronger code with a lower code rate.
Source decoded output has artefacts (audio glitches, video blocking, corrupted data)
Cause: Uncorrectable errors reaching the source decoder due to insufficient FEC protection, or a mismatch between the compressed stream format produced by the encoder and the format expected by the decoder Fix: Verify the channel codec is providing adequate error protection for the operating BER. Check that encoder and decoder codec versions and parameter sets are compatible. Implement packet loss concealment or error resilience tools at the source codec level (frame refresh in video, packet loss concealment in audio).

Frequently asked questions

What are the main blocks in a digital communication system?

The main blocks are: (1) Information source; (2) Source encoder (compression); (3) Channel encoder (error correction coding); (4) Modulator (maps bits to waveform); (5) Transmission channel (physical medium with noise); (6) Demodulator (recovers bits from received signal); (7) Channel decoder (error correction); (8) Source decoder (decompression); (9) Destination (user or device).

What is the difference between source coding and channel coding?

Source coding removes redundancy from the information to reduce the number of bits required — it compresses data. Channel coding adds structured redundancy back to the bit stream to enable error detection and correction at the receiver. The two operations are applied in sequence and serve opposite goals, both essential for efficient and reliable communication.

What is AWGN and why is it used to model communication channels?

AWGN stands for Additive White Gaussian Noise. It is a mathematical model where noise of uniform power spectral density (white) is added to the signal and follows a Gaussian amplitude distribution. Real thermal noise in electronic receivers closely follows this model. The AWGN model is used to derive analytical performance bounds and compare modulation and coding schemes using the Eb/N0 ratio (energy per bit to noise spectral density).

What is the Shannon–Hartley theorem and how does it relate to this block diagram?

The Shannon–Hartley theorem defines the maximum error-free bit rate (channel capacity C) achievable over a channel with a given bandwidth B and signal-to-noise ratio SNR: C = B × log₂(1 + SNR). It sets the theoretical upper limit on what modulation and coding schemes combined can achieve — the combined source encoder, channel encoder, and modulator must operate below this limit to enable reliable communication.

What is forward error correction (FEC) and when is it used?

FEC is a channel coding technique where the transmitter adds redundant check bits derived from the data bits. The receiver uses these check bits to detect and correct errors without requesting retransmission. FEC is essential where retransmission is impractical — satellite links (very long round-trip delay), broadcast systems (no return path), and real-time audio/video. Examples include Reed–Solomon codes (DVB, Blu-ray), turbo codes (3G/4G), and LDPC codes (5G, Wi-Fi 6).

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