Introduction / Context:
Real-world phenomena like sound are analog: they vary continuously over time. Digital systems sample and quantize these signals, storing and processing them as sequences of numbers. The question asks if digital audio recording exemplifies this concept—converting analog music into a digital form for storage and playback.
Given Data / Assumptions:
- Analog audio is captured by a microphone producing a voltage proportional to air pressure variations.
- An ADC performs sampling and quantization to produce digital samples.
- Common formats: PCM at sample rates like 44.1 kHz, 48 kHz, and bit depths like 16-bit, 24-bit.
Concept / Approach:
Sampling captures the signal at discrete times; quantization maps amplitudes to finite levels. For faithful reconstruction, the sampling theorem recommends a rate exceeding twice the highest frequency of interest. Digital storage allows robust editing, compression, and error correction while maintaining consistent playback quality, provided the digital-to-analog conversion and output chain are competent.
Step-by-Step Solution:
Analog music → microphone converts pressure to voltage.ADC samples at a chosen rate and quantizes to N bits per sample.Data is stored/processed digitally (files, streams, DSP).DAC reconverts samples to analog for speakers or headphones.
Verification / Alternative check:
Listen for consistent quality across copies; digital cloning preserves the exact sample values unlike analog tape dubbing.
Why Other Options Are Wrong:
Incorrect: Digital audio is the textbook example of digitized analog signals.True only for speech: Music and speech are both analog waveforms suited to digitization.Applies only above 48 kHz: Valid digital audio exists at many rates (e.g., 44.1 kHz CDs).
Common Pitfalls:
Confusing sampling rate with bit depth; both affect fidelity differently.Assuming compressed formats (MP3/AAC) are identical to PCM; they are perceptually coded but still digital representations.
Final Answer:
Correct
Discussion & Comments