Sound & Audio Processing: Speaking and Listening Focus | Edexcel 计算机科学：口语与听力数字处理备考专项

📚 Sound & Audio Processing: Speaking and Listening Focus | 口语与听力数字处理备考专项

In Edexcel GCSE Computer Science, we often think about how computers ‘speak’ and ‘listen’ – not with human voices, but through speakers and microphones. This revision guide focuses on the digital processing of sound, covering sampling, file sizes, compression and the key principles that let machines capture and reproduce audio. Mastering these topics will prepare you for the data representation questions that regularly appear on Paper 1 and Paper 2.

在 Edexcel GCSE 计算机科学中，我们经常需要思考计算机如何“说”和“听”——不是用人类的声音，而是通过扬声器和麦克风。本复习指南专注于声音的数字处理，包含采样、文件大小计算、压缩以及让机器捕捉并还原音频的关键原理。掌握这些内容能帮助你应对试卷一和试卷二中经常出现的数据表示题目。

1. Introduction to Sound in Computer Science | 计算机科学中的声音简介

Sound is a vital form of data. Computers use audio for everything from voice assistants and notifications to music streaming and podcasts. To store or transmit sound, a computer must convert continuous physical sound waves into binary digits. This process involves hardware components that act as ‘ears’ (microphones) and ‘mouths’ (speakers).

声音是一种重要的数据形式。从语音助手、通知提示到音乐串流和播客，计算机在各种场景中使用音频。为了储存或传输声音，计算机必须将连续的物理声波转换为二进制数字。这一过程涉及扮演“耳朵”（麦克风）和“嘴巴”（扬声器）的硬件组件。

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2. Analogue vs Digital Sound | 模拟声音与数字声音

The sound we hear in the real world is analogue – a smooth, continuously varying waveform. Computers, however, operate with discrete binary values. Analogue signals must be digitised before a computer can process them. The conversion from analogue to digital (ADC) and digital to analogue (DAC) is fundamental to all audio technology.

我们听到的真实世界声音是模拟的——一种平滑、连续变化的波形。然而，计算机使用离散的二进制值进行处理。模拟信号必须先被数字化，计算机才能处理它。模数转换 (ADC) 和数模转换 (DAC) 是所有音频技术的基础。

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3. The Microphone: Listening to Sound | 麦克风：听声音

A microphone acts as the computer’s listening device. It captures analogue sound waves and sends them to an Analogue-to-Digital Converter (ADC). The ADC samples the analogue signal at regular intervals and produces a stream of binary numbers. Without this ‘listening’ step, a computer would have no way to receive spoken commands or record music.

麦克风充当计算机的收听设备。它捕捉模拟声波，并将其传送至模数转换器 (ADC)。ADC 按固定间隔对模拟信号进行采样，产生一连串二进制数字。没有这个“听”的步骤，计算机就无法接收语音指令或录制音乐。

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4. The Speaker: Speaking Sound | 扬声器：说声音

To reproduce sound, a computer uses a Digital-to-Analogue Converter (DAC). The DAC reads binary audio data and reconstructs an analogue waveform, which is then amplified and sent to a speaker. The speaker diaphragm vibrates to create sound waves that our ears can detect. This is how a computer ‘speaks’ – by transforming numbers back into pressure waves.

为了重现声音，计算机使用数模转换器 (DAC)。DAC 读取二进制音频数据，重建模拟波形，经放大后发送给扬声器。扬声器振膜振动产生我们能听到的声波。这就是计算机“说话”的方式——把数字变回气压波。

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5. Sampling: Converting Sound to Data | 采样：将声音转换为数据

Sampling is the process of measuring the amplitude of an analogue sound wave at equally spaced time intervals. Each measurement is called a sample. The continuous waveform is only recorded at specific points, which means some information is lost – but if we sample often enough, the digital copy sounds nearly identical to the original.

采样是指在等距的时间间隔上测量模拟声波振幅的过程。每次测量称为一个样本。连续波形只在特定点上被记录下来，这意味着会有部分信息丢失——但如果采样频率足够高，数字副本听起来几乎与原声一致。

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6. Sample Rate and Bit Depth | 采样率与位深度

Two parameters control the quality of digitised sound. Sample rate is the number of samples taken per second, measured in Hertz (Hz). A typical CD uses 44 100 Hz (44.1 kHz). Bit depth is the number of bits used to store each sample; commonly 16 bits on a CD. Higher sample rates capture higher frequencies, while greater bit depth gives finer amplitude resolution.

两个参数控制数字化声音的质量。采样率是每秒采样的次数，单位为赫兹 (Hz)。典型 CD 使用 44 100 Hz (44.1 kHz)。位深度是储存每个样本所用的位数，CD 通常为 16 位。更高的采样率能捕捉更高的频率，更大的位深度则提供更精细的振幅分辨率。

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7. Nyquist Theorem | 奈奎斯特定理

The Nyquist theorem states that to accurately reproduce a sound, the sample rate must be at least twice the highest frequency present in the original signal. Human hearing ranges up to about 20 kHz, so a sample rate of at least 40 kHz is needed. The CD standard of 44.1 kHz satisfies this requirement. If you sample too slowly, aliasing occurs – false frequencies appear in the reconstruction.

奈奎斯特定理指出，要准确还原声音，采样率必须至少是原始信号中最高频率的两倍。人类听力范围大约可达 20 kHz，因此需要至少 40 kHz 的采样率。CD 标准的 44.1 kHz 满足这一要求。如果采样过慢，会发生混叠——重建信号中出现虚假频率。

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8. Calculating Audio File Size | 计算音频文件大小

Exam questions often ask you to calculate the size of an uncompressed audio file. The formula is:

考试题目经常要求计算未压缩音频文件的大小。公式如下：

File size (bits) = sample rate (Hz) × bit depth × duration (seconds) × number of channels

Remember to convert bits to bytes by dividing by 8, and to kilobytes by dividing by 1024 if required. Be careful with units – always show your working clearly.

切记，将位转换为字节需除以 8，需要时再除以 1024 得到千字节。注意单位换算——务必清晰地展示运算步骤。

Example: Calculate the file size in MB for a 3-minute stereo track recorded at 44.1 kHz with 16-bit depth. Stereo = 2 channels.

示例：计算一段 3 分钟立体声轨道，以 44.1 kHz、16 位深度录制时的文件大小，单位为 MB。立体声 = 2 声道。

Size = 44 100 × 16 × (3 × 60) × 2 = 44 100 × 16 × 180 × 2 = 253 440 000 bits

253 440 000 ÷ 8 = 31 680 000 bytes
31 680 000 ÷ 1024 ≈ 30 937.5 KB
30 937.5 ÷ 1024 ≈ 30.2 MB

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9. Audio Compression: Lossy vs Lossless | 音频压缩：有损与无损

Uncompressed audio files can be very large, so compression is often used. Lossless compression (e.g., FLAC) reduces file size without removing any audio data – the original can be perfectly reconstructed. Lossy compression (e.g., MP3, AAC) permanently removes sounds the human ear is unlikely to notice, achieving much smaller files but with some loss of quality.

未压缩的音频文件可能十分庞大，因此常使用压缩。无损压缩（如 FLAC）减小文件体积但不移除任何音频数据——原文件可被完美重建。有损压缩（如 MP3、AAC）会永久性地移除人耳不易察觉的声音，从而获得更小的文件，但会牺牲部分音质。

Property	Lossless	Lossy
Quality preservation	Perfect	High, but not identical
Typical file size reduction	30–50 %	75–95 %
Common formats	FLAC, ALAC	MP3, AAC, OGG

Knowing when to use each type is a common multiple-choice topic. For streaming, lossy is preferred because of bandwidth limits; for archiving, lossless keeps a perfect copy.

了解何时使用每种压缩是常见的选择题考点。对于串流传输，因带宽限制，优先选择有损格式；为存档保留完美副本，则使用无损格式。

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10. Practical Example: Recording a Voice Note | 实例：录制语音备忘录

Imagine recording a 10-second voice note on a smartphone. The microphone picks up analogue sound; the ADC samples it at, say, 8 kHz with an 8-bit depth for a mono, voice-grade recording. The raw file size would be:

想象在智能手机上录制一段 10 秒的语音备忘录。麦克风拾取模拟声音；ADC 以比如 8 kHz 采样率、8 位深度进行单声道语音级录制。原始文件大小为：

8 000 × 8 × 10 × 1 = 640 000 bits = 80 000 bytes ≈ 78.1 KB

If the app then applies lossy compression, the final file might be only 5–10 KB, which saves storage space while keeping the speech perfectly intelligible. This shows the complete ‘listening and speaking’ cycle: the computer listens via the mic, converts to binary, and can later ‘speak’ the stored message through a speaker.

如果应用随后采用有损压缩，最终文件可能只有 5–10 KB，在节省储存空间的同时仍能保持语音的清晰可懂。这展示了完整的“听与说”循环：计算机通过麦克风“听”，转换为二进制，之后可通过扬声器“说出”存储的信息。

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11. Review and Key Formulas | 复习与关键公式

Let’s consolidate the core relationships you must recall in the exam. The most important formula is the uncompressed file size calculation. Also remember the Nyquist requirement for correct reconstruction. Repeated practice with unit conversions will build speed.

下面巩固你在考试中必须记住的核心关系。最重要的公式就是未压缩文件大小计算。同时要记住为正确还原所需的奈奎斯特条件。反复练习单位换算可提升答题速度。

• Uncompressed file size (bits) = sample rate × bit depth × time × channels
• Uncompressed file size (bits) = sample rate × bit depth × time × channels
• Minimum sample rate = 2 × highest frequency (Nyquist)
• 最低采样率 = 2 × 最高频率 (奈奎斯特)
• 1 byte = 8 bits; 1 KB = 1024 bytes; 1 MB = 1024 KB
• 1 字节 = 8 位；1 KB = 1024 字节；1 MB = 1024 KB

Keep a careful eye on whether the question expects the answer in bits, bytes, kilobytes or megabytes. Underline the unit in the question before you start calculating.

仔细看清题目要求答案的单位是位、字节、千字节还是兆字节。开始计算前先将单位下划线标出。

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12. Exam Tips for Sound Processing Questions | 声音处理题目的考试技巧

Sound representation questions appear in both multiple-choice and structured formats. Show every step of your working – examiners award marks for correct method even if the final number is off. Check that your sample rate and time units match (seconds, not minutes). If given a frequency range, apply Nyquist immediately. For comparisons between compressed and uncompressed audio, mention storage implications, data loss and bandwidth.

声音表示题目会以单选和结构化题形式出现。务必展示每一步运算过程——即使最终数字有误，正确的计算方法仍能得分。检查采样率与时间单位是否一致（使用秒，不是分钟）。如果给出频率范围，立即应用奈奎斯特定理。对比压缩与未压缩音频时，要提及储存空间影响因素、数据丢失以及带宽问题。

A common pitfall is forgetting to multiply by the number of channels. A stereo signal has two channels, so the raw bit rate doubles. Practice with mono, stereo and surround sound examples to build confidence.

一个常见陷阱是忘记乘以声道数。立体声信号有两个声道，因此原始比特率翻倍。通过单声道、立体声和环绕声示例进行练习，可树立自信。

Published by TutorHao | Computer Science Revision Series | aleveler.com

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