📝 Streaming stored audio video in computer networks (26 MCQs)
📖 From Data Communication and Networks • 28. Multimedia • 26 questions available
What is Streaming stored audio video in computer networks?
Streaming stored audio video in computer networks involves transmitting pre-recorded media files from a server to clients using progressive download or adaptive bitrate streaming, allowing playback to begin before the entire file is downloaded while adjusting quality dynamically based on available bandwidth to minimize buffering and ensure smooth viewing experiences.
📝 All Streaming stored audio video in computer networks MCQs
Q1. What does the term 'streaming' refer to in the context of stored audio/video?
📖 Explanation: Streaming means sending media data in a continuous flow so the receiver can begin playback before the entire file arrives, which distinguishes it from traditional download where the whole file must be stored first.
Q2. Which of the following is a common compression method used for streaming video?
📖 Explanation: MPEG-4 is widely adopted for streaming because it provides efficient compression while preserving visual quality, making it suitable for bandwidth‑limited networks. JPEG is for still images, MP3 for audio, and WAV is an uncompressed audio format.
Q3. In run-length encoding for binary patterns, what does a 'digit' represent when using 4 bits per digit?
📖 Explanation: When encoding a binary stream with RLE, each digit stores the count of zeros that appear between successive ones. Using a fixed 4‑bit field limits the count to a maximum of 15, and the digit does not convey any other information.
Q4. If a streaming server increases its buffer size, what is the most likely effect on playback smoothness under variable network conditions?
📖 Explanation: A larger buffer absorbs short‑term bandwidth fluctuations, allowing the player to continue feeding data to the decoder even when packets arrive late, which improves smoothness at the cost of a longer initial delay.
Q5. Given a binary pattern 1 0 0 1 0 1, after applying run-length encoding with count of 0s between 1s, what is the resulting sequence?
📖 Explanation: The first two 1s have two zeros between them, giving a count of 2. Between the second and third 1 there is one zero, giving a count of 1. The encoded sequence is therefore \2
Q6. If a codec compresses audio at 128 kbps and the network bandwidth drops to 64 kbps, which outcome is most probable?
📖 Explanation: When the available bandwidth falls below the required bitrate, the player cannot retrieve data fast enough, leading to buffer depletion and audible glitches or stutter until the bandwidth recovers.
Q7. Assume a streaming protocol uses a fixed 4‑bit field to encode the count of zeros between ones. What is the maximum run length of zeros that can be represented?
📖 Explanation: A 4‑bit unsigned field can represent values from 0 to 15 inclusive, so the longest run of zeros that can be stored is fifteen zeros.
Q8. When streaming stored video using progressive download, which of the following best explains why initial buffering reduces start‑up delay compared to true streaming?
📖 Explanation: Buffering stores enough decoded frames to keep the decoder fed, so playback can start after a short pause while the rest of the file continues downloading in the background.
Q9. If a streaming service switches from constant bitrate (CBR) to variable bitrate (VBR) encoding, what is the most direct impact on bandwidth usage?
📖 Explanation: VBR adjusts the amount of data per unit time based on content complexity, so the required bandwidth can rise during complex scenes and fall during simpler ones, leading to variable usage.
Q10. During streaming, packet loss typically leads to which immediate effect on audio quality?
📖 Explanation: Lost packets mean missing audio samples; most players insert silence or repeat previous samples, causing audible gaps or glitches that are perceived as brief interruptions.
Q11. Compare run-length encoding (RLE) with Huffman coding for binary data with many zeros. Which statement is accurate?
📖 Explanation: RLE compresses long sequences of identical symbols efficiently, so when zeros appear in long runs it often yields better compression than Huffman, which optimizes symbol frequencies but may not capture run length patterns.
Q12. Evaluate the trade‑off between increasing the bits per digit in RLE (e.g., from 4 to 8 bits) and the resulting compression ratio for a binary pattern with average run length of 10 zeros. Which outcome is most likely?
📖 Explanation: Using more bits per digit adds overhead for each run count; when the average run length is modest (10), the extra bits outweigh the benefit, so the overall compression ratio gets worse.
Q13. Which scenario best illustrates the advantage of using streaming over downloading for live events?
📖 Explanation: Live streaming delivers content as it happens, enabling viewers to watch a sports match with only a short delay, whereas downloading would require the entire event to finish before playback could begin.
Q14. Contrast the latency characteristics of HTTP progressive download versus RTSP streaming. Which statement is correct?
📖 Explanation: RTSP is designed for real‑time delivery and can start playback with minimal buffering, whereas HTTP progressive download often waits for a larger buffer, resulting in higher start‑up latency.
Q15. When adapting streaming quality for mobile users, adaptive bitrate (ABR) algorithms must balance which two primary factors?
📖 Explanation: ABR monitors how much data is currently buffered and estimates the available network bandwidth, then selects a video bitrate that keeps the buffer from emptying while not exceeding the link capacity.
Q16. In the context of streaming stored video, how does interleaving of I‑frames, P‑frames, and B‑frames affect error resilience?
📖 Explanation: I‑frames are self‑contained; increasing their frequency means that when a corrupted frame occurs, the decoder can recover more quickly, improving resilience to transmission errors.
Q17. Which of the following best differentiates 'buffering' from 'pre‑fetching' in streaming terminology?
📖 Explanation: Buffering refers to temporarily holding incoming data during playback, while pre‑fetching deliberately downloads data ahead of time so that playback can start immediately when the user requests it.
Q18. Consider two streaming protocols: one uses TCP and the other uses UDP. Which protocol is more suitable for real‑time video where occasional loss is acceptable?
📖 Explanation: UDP provides low‑latency delivery without retransmission, making it preferable for live video where a few lost packets are tolerable, whereas TCP's reliability mechanisms add delay unsuitable for real‑time streams.
Q19. If a binary audio sample stream is encoded using RLE with a 4‑bit count field, how many distinct run lengths can be represented?
📖 Explanation: A 4‑bit unsigned field can encode 2⁴ = 16 different values, representing run lengths from 0 up to 15 inclusive.
Q20. Apply the principle of temporal locality to explain why increasing the initial buffer size can reduce playback interruptions.
📖 Explanation: Temporal locality means that data needed soon is likely to have been accessed recently; a larger initial buffer holds the most recent samples, so brief network slowdowns do not immediately cause gaps.
Q21. Synthesize a method to combine run‑length encoding with Huffman coding for binary streams to achieve better compression. Which step order is most effective?
📖 Explanation: First applying RLE reduces long runs to short count symbols; subsequently Huffman coding can efficiently encode the resulting symbol frequencies, yielding better overall compression than either method alone.
Q22. Explain how adaptive streaming leverages client‑side bandwidth estimation to select appropriate quality levels.
📖 Explanation: The client continuously monitors download speed, then requests video segments encoded at a bitrate that matches the measured throughput, enabling smooth playback while avoiding rebuffering.
Q23. Given a streaming scenario where packet loss probability is 0.01 and each lost packet requires retransmission, estimate the expected additional latency per 1000 packets assuming a round‑trip time of 50 ms.
📖 Explanation: Out of 1000 packets, 0.01 × 1000 = 10 are lost; each loss adds a round‑trip time of 50 ms, so total extra latency is 10 × 50 = 500 ms, which is the expected additional delay.
Q24. Which principle explains why audio codecs often use psychoacoustic models to discard inaudible frequencies?
📖 Explanation: Psychoacoustic models exploit the masking effect, where louder sounds make nearby frequencies imperceptible, allowing codecs to remove those inaudible components without noticeably degrading perceived quality.
Q25. In streaming stored video, how does keyframe interval selection affect seekability?
📖 Explanation: Keyframes (I‑frames) are points where decoding can start without prior frames; having them more frequently gives the player more locations to jump to, making random seeking faster and more precise.
Q26. Apply the concept of buffering to explain why live streaming of a sports event may exhibit a few seconds of delay compared to the actual event.
📖 Explanation: A small buffer smooths out variations in packet arrival time (jitter) and provides enough data for continuous decoding, which inevitably introduces a short latency between the live event and what the viewer sees.