pulse code

简明释义

脉码

英英释义

例句

1.The video compression algorithm relies on pulse code modulation to reduce file size.

视频压缩算法依赖于脉冲编码调制来减小文件大小。

2.When sending data, the system converts it into pulse code for efficient transmission.

在发送数据时，系统将其转换为脉冲编码以实现高效传输。

3.In telecommunications, a pulse code is essential for transmitting voice signals over long distances.

在电信中，脉冲编码对于远距离传输语音信号至关重要。

4.The audio engineer used a pulse code to digitize the sound waves for the music track.

音频工程师使用脉冲编码将声音波形数字化以制作音乐轨道。

5.Digital signal processors often utilize pulse code techniques for audio processing.

数字信号处理器通常利用脉冲编码技术进行音频处理。

作文

In the realm of digital communication and audio processing, the term pulse code refers to a method used to digitally represent analog signals. This technique is fundamental in various applications, including telecommunication and audio recording. The basic idea behind pulse code modulation (PCM) is to sample an analog signal at regular intervals and convert these samples into a sequence of coded pulses. This process allows for high fidelity in sound reproduction and efficient transmission of audio data.To understand pulse code more clearly, we can break it down into several key components. First, sampling is the process of measuring the amplitude of an analog signal at discrete time intervals. The Nyquist theorem states that to accurately reconstruct a signal, it must be sampled at least twice its highest frequency. For instance, if an audio signal contains frequencies up to 20 kHz, it should be sampled at least at 40 kHz.Once the signal has been sampled, the next step involves quantization. This is where the continuous amplitude values are converted into discrete levels. Each sampled value is assigned a binary number based on its amplitude. The more bits used in this representation, the finer the resolution of the quantized signal. For example, using 8 bits allows for 256 different amplitude levels, while 16 bits provides 65,536 levels, resulting in a more accurate representation of the original signal.After quantization, the final step in the pulse code process is encoding. In this phase, the quantized values are converted into a binary format that can be easily transmitted or stored. This encoded information consists of a series of '1's and '0's, which can be processed by digital systems and transmitted over various media.The advantages of using pulse code modulation are numerous. One significant benefit is the ability to transmit audio signals over long distances without degradation. Unlike analog signals, which can suffer from noise and distortion, PCM maintains the integrity of the original signal, ensuring high-quality playback. Additionally, PCM is compatible with digital storage formats, making it ideal for modern audio applications such as CDs and digital streaming.However, there are also some challenges associated with pulse code modulation. The primary concern is the bandwidth required for transmission. Since PCM involves high sampling rates and bit depths, it can consume a considerable amount of bandwidth, which may not be feasible in all situations. This limitation has led to the development of various compression techniques, such as MP3 and AAC, which aim to reduce the size of audio files while preserving sound quality.In conclusion, pulse code modulation is a crucial technology in the field of digital audio and communication. Its ability to accurately represent analog signals in a digital format has revolutionized how we record, transmit, and reproduce sound. By understanding the principles of sampling, quantization, and encoding, we can appreciate the significance of pulse code in our daily lives, from listening to music on our devices to communicating over the phone. As technology continues to evolve, the importance of pulse code will undoubtedly remain at the forefront of digital audio advancements.

在数字通信和音频处理领域，术语脉冲编码指的是一种用于数字表示模拟信号的方法。这种技术在各种应用中都是基础，包括电信和音频录制。脉冲编码调制（PCM）的基本思想是以规则的间隔对模拟信号进行采样，并将这些样本转换为一系列编码脉冲。这一过程使得声音重现的高保真度和音频数据的高效传输成为可能。为了更清楚地理解脉冲编码，我们可以将其分解为几个关键组成部分。首先，采样是以离散时间间隔测量模拟信号幅度的过程。奈奎斯特定理指出，为了准确重建信号，必须以至少其最高频率的两倍进行采样。例如，如果音频信号包含高达20 kHz的频率，则应至少以40 kHz进行采样。一旦信号被采样，下一步就是量化。在这一阶段，连续的幅度值被转换为离散级别。每个采样值根据其幅度分配一个二进制数字。用于这种表示的位数越多，量化信号的分辨率就越细。例如，使用8位可以提供256个不同的幅度级别，而16位则提供65,536个级别，从而更准确地表示原始信号。在量化之后，脉冲编码过程的最后一步是编码。在这一阶段，量化值被转换为易于传输或存储的二进制格式。这些编码信息由一系列的'1'和'0'组成，可以被数字系统处理并通过各种媒体传输。使用脉冲编码调制的优点有很多。一个显著的好处是能够在长距离内传输音频信号而不失真。与模拟信号不同，模拟信号可能会受到噪声和失真的影响，而PCM保持了原始信号的完整性，确保高质量的播放。此外，PCM与数字存储格式兼容，使其成为现代音频应用的理想选择，例如CD和数字流媒体。然而，脉冲编码调制也面临一些挑战。主要问题是传输所需的带宽。由于PCM涉及高采样率和位深度，因此可能会消耗相当大的带宽，这在某些情况下可能不可行。这一限制促使各种压缩技术的发展，例如MP3和AAC，旨在在保留音质的同时减少音频文件的大小。总之，脉冲编码调制是数字音频和通信领域的一项关键技术。它能够准确地以数字格式表示模拟信号，彻底改变了我们记录、传输和重现声音的方式。通过理解采样、量化和编码的原理，我们可以欣赏到脉冲编码在我们日常生活中的重要性，从在设备上听音乐到通过电话进行交流。随着技术的不断发展，脉冲编码的重要性无疑将继续处于数字音频进步的前沿。