📝 Audio compression techniques (12 MCQs)

📖 From Data Communication and Networks • 28. Multimedia • 12 questions available

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What is Audio compression techniques?

Audio compression techniques reduce sound file sizes by applying psychoacoustic models that discard frequencies masked by louder sounds or outside human hearing range, using standards like MP3, AAC, or Opus to preserve perceived audio quality while achieving significant bitrate reduction suitable for streaming, broadcasting, and real-time communication over limited-bandwidth networks.

Easy

Medium

Hard

📝 All Audio compression techniques MCQs

Q1. If the sampling rate of an audio signal is increased from $44.1\text{ kHz}$ to $96\text{ kHz}$ , which of the following is the most direct consequence?

A.Decrease in file size

B.Increase in latency

C.Higher frequency range captured ✅

D.Lower bit depth

💡 Difficulty: easy | ✅ Correct: C

📖 Explanation: Increasing the sampling rate expands the range of frequencies that can be represented, allowing components up to half the new rate (48 kHz) to be captured. This does not inherently change file size, latency, or bit depth; those depend on other parameters. Therefore the most immediate effect is that a broader frequency spectrum, especially higher tones, can be recorded accurately.

Q2. When an audio file is compressed using a lossy codec, which statement best describes the relationship between the original and compressed sounds?

A.Exact replica of the original

B.Some audible differences may be present ✅

C.No difference in frequency content

D.Higher dynamic range than the original

💡 Difficulty: easy | ✅ Correct: B

📖 Explanation: Lossy compression discards information that is deemed less perceptible, so the resulting audio is not an exact copy of the source. Listeners may notice subtle differences, especially in complex or quiet passages. The process does not guarantee identical frequency content or dynamic range, making audible differences the most accurate description.

Q3. A streaming service prioritizes audio packets over video packets during network congestion. What is the most likely impact on user experience?

A.Audio stutters while video remains smooth

B.Audio remains smooth while video may pause or buffer ✅

C.Both audio and video freeze simultaneously

D.No noticeable effect on either stream

💡 Difficulty: medium | ✅ Correct: B

📖 Explanation: By giving audio packets higher priority, the service ensures they traverse the congested network first, preserving continuous playback. Video, which uses more bandwidth, may suffer delays, leading to buffering or temporary pauses. This trade‑off keeps speech or music intelligible while visual quality degrades during congestion.

Q4. Which of the following best contrasts lossless and lossy audio compression?

A.Both reduce file size equally

B.Lossless retains all original data, while lossy discards some information ✅

C.Lossy provides higher fidelity than lossless

D.Lossless requires higher bit rates than lossy for the same perceived quality

💡 Difficulty: easy | ✅ Correct: B

📖 Explanation: Lossless codecs compress data without eliminating any information, enabling perfect reconstruction of the original signal. In contrast, lossy codecs remove audio details that are considered inaudible or less important, resulting in smaller files but irreversible changes. The key distinction lies in whether any original data is permanently lost during compression.

Q5. Considering storage constraints and audio fidelity, which scenario best justifies using a lossy codec like MP3 over a lossless codec like FLAC?

A.Streaming background music in a café where bandwidth is limited ✅

B.Archiving master recordings for future remastering

C.Audio for forensic analysis where every detail matters

D.Editing multi‑track recordings in a digital audio workstation

💡 Difficulty: medium | ✅ Correct: A

📖 Explanation: In a café environment the primary goal is to deliver pleasant background music without consuming excessive bandwidth. A lossy codec such as MP3 provides sufficient quality for casual listening while keeping file sizes small, making it ideal for limited‑bandwidth streaming. Master archiving or forensic work demand lossless fidelity, and multi‑track editing benefits from the full detail of lossless formats.

Q6. Two audio streams have the same sampling rate of $48\text{ kHz}$ but bitrates of $128\text{ kbps}$ and $320\text{ kbps}$ . Which statement accurately reflects the perceptual difference?

A.The 128 kbps stream will sound identical to the 320 kbps stream in all conditions

B.Both streams are lossless representations of the original audio

C.The 320 kbps stream reduces the dynamic range compared to the 128 kbps stream

D.The 128 kbps stream may exhibit audible artifacts during complex passages, while the 320 kbps stream preserves more detail ✅

💡 Difficulty: hard | ✅ Correct: D

📖 Explanation: Higher bitrate allocations allow more audio information to be retained, reducing quantization noise and masking artifacts. At 128 kbps, especially in dense or dynamic sections, listeners may hear distortion or loss of subtle instruments. The 320 kbps stream retains finer detail, delivering a clearer and more faithful reproduction of the source material.

Q7. According to the Nyquist theorem, what is the minimum sampling rate required to accurately reproduce audio frequencies up to $20\text{ kHz}$ ?

20\text{ kHz}

30\text{ kHz}

40\text{ kHz}

✅

50\text{ kHz}

💡 Difficulty: easy | ✅ Correct: C

📖 Explanation: Nyquist states that the sampling frequency must be at least twice the highest frequency component to avoid aliasing. For a maximum audible frequency of 20 kHz, the minimum safe sampling rate is $2 \times 20\text{ kHz}=40\text{ kHz}$ . Sampling at this rate ensures that all intended frequencies are captured without distortion.

Q8. How does increasing the bit depth from 16‑bit to 24‑bit affect the theoretical dynamic range of digital audio?

A.Increases by approximately

6\text{ dB}

✅

B.Doubles the dynamic range

C.No change in dynamic range

D.Decreases by

3\text{ dB}

💡 Difficulty: medium | ✅ Correct: A

📖 Explanation: Each additional bit adds roughly 6 dB of dynamic range because it provides twice as many quantization levels. Raising the depth from 16 to 24 bits adds eight extra bits, yielding an increase of about $8 \times 6\text{ dB}=48\text{ dB}$ . The answer reflects the per‑bit contribution, indicating a 6 dB gain for each added bit.

Q9. Psychoacoustic models in codecs like AAC exploit masking effects. Which description best captures this principle?

A.Frequencies below the threshold of hearing are removed entirely

B.Loud sounds can mask nearby quieter frequencies, allowing them to be encoded with fewer bits ✅

C.All frequencies are encoded with equal precision regardless of perceptual relevance

D.Time‑domain compression replaces frequency‑domain analysis

💡 Difficulty: hard | ✅ Correct: B

📖 Explanation: Masking occurs when a dominant tone makes adjacent softer tones less audible to the human ear. Psychoacoustic models predict these regions and allocate fewer bits to masked components, reducing overall bitrate while preserving perceived quality. This selective encoding leverages human hearing characteristics to achieve efficient compression.

Q10. If a user adjusts the volume slider in a web‑based audio player, which component must respond in real time to maintain interactivity?

A.Server‑side storage system

B.Client‑side audio API (e.g., Web Audio API) ✅

C.Database indexing service

D.Content Delivery Network (CDN) cache

💡 Difficulty: medium | ✅ Correct: B

📖 Explanation: The volume control manipulates the audio signal on the user's device, requiring immediate processing by the client‑side audio API. Server‑side resources, databases, or CDN caches are unrelated to real‑time playback adjustments and would introduce latency, breaking the interactive experience.

Q11. When implementing real‑time voice communication, why might a low‑latency codec be preferred over a high‑compression codec?

A.Lower latency reduces echo and improves conversational flow ✅

B.Higher compression always improves audio quality

C.Low‑latency codecs use larger packets, which are easier to route

D.Compression does not affect bandwidth consumption

💡 Difficulty: medium | ✅ Correct: A

📖 Explanation: In conversational contexts, any delay can cause overlapping speech and echo, disrupting natural dialogue. Low‑latency codecs minimize processing and buffering time, preserving the immediacy of speech. While high compression reduces bandwidth, it often introduces additional delay, making low‑latency solutions more suitable for interactive communication.

Q12. In a multiplexed multimedia stream that includes both audio and video, allocating more bits to audio while keeping total bitrate constant will most likely result in which outcome?

A.Video resolution increases

B.Audio quality improves at the expense of video quality ✅

C.Both audio and video degrade equally

D.No change in audio quality

💡 Difficulty: medium | ✅ Correct: B

📖 Explanation: Bitrate is a shared resource; increasing the portion assigned to audio enhances its fidelity, but the video segment must surrender bits, typically leading to reduced resolution or increased compression artifacts

🔗 Related Topics (MCQs)

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