Text 3: Digital Audio

In the early computer, the only sound that we heard from a computer was a beep—often accompanied by an error message. Now an entire range of sounds can be played through the computer, including music, narration, sound effects, and original recording of events such as a speech or a concert. The elements of sound used in computer often called digital audio, and it is fundamental to multimedia.

Now we have a rough idea of what's happening when we hear a sound, we can begin to make sense of what audio experts cryptically refer to as “waveform” diagrams. Here's what a waveform looks like (Figure 3).

Waveforms aren't as intimidating as they look. In essence, a waveform is a graph that charts minute changes in air pressure as sound waves propagate. The y axis represents air pressure and the x axis represents time (Figure 4).

A sound wave arrives. The air pressure goes up and the line on the waveform rises. As the sound wave passes the air pressure falls again and the line falls as well. These changes in pressure happen very quickly—thousands of times as a second. Waveform diagrams representing even a few seconds of sound are, consequently, very big. This is the reason why waveforms of audio recordings often look complicated and squiggly when you view them on your computer.

There are several technical terms used to describe waveforms (Figure 5) that you should know. They'll come into use when we get to discussing digital audio.

Zero Line: The horizontal line running through the middle of the graph is called the zero line. It represents the rest state, when there is no compression or rarefaction.

Amplitude: Amplitude is the amount of compression or rarefaction at any point on the waveform. Graphically, it is the distance above or below the zero line. In general, the greater the amplitude is, the louder the volume is.

Cycle: A cycle is the amount of time it takes for the amplitude of the waveform to return to the same level.

Frequency: The frequency of a sound is the number of cycles that happen every second. The higher is the frequency, the higher is the perceived pitch of a sound. For humans, hearing is limited to frequencies between about 20 Hz and 20, 000Hz (20 kHz), with the upper limit generally decreasing with age. Other species have a different rage of hearing. The average dog can hear frequencies as high as 45,000 Hz. Cats can hear up to 63,000Hz, and the beluga whale can hear frequencies of up to 123, 000Hz.

Digital audio has emerged because of its usefulness in the recording, manipulation, mass—production, and distribution of sound. Modern distribution of music across Internet through on-line stores depends on digital recording and digital compression algorithms. Distribution of audio as data files rather than as physical objects has significantly reduced cost of distribution.

Digital audio uses digital signals for sound reproduction. This includes analog-to-digital conversion, digital-to-analog conversion, storage, and transmission.

A digital audio signal starts with an analog-to-digital converter (ADC) that converts an analog signal to a digital signal. The ADC runs at a sampling rate and converts at a known bit resolution. For example, CD audio has a sampling rate of 44.1 kHz (44,100 samples per second) and 16-bit resolution for each channel (stereo). If the analog signal is not already band limited then an anti-aliasing filter is necessary before conversion, to prevent aliasing in the digital signal.

Sound waves ripple past the microphone, causing a diaphragm inside to vibrate. The vibrating diaphragm creates a change in voltage in the wire that runs from the microphone to the computer. This fluctuating voltage is an analog representation of the sound, for it changes smoothly from one amplitude to the next, encompassing all values in-between. Inside of our computer the ADC (usually a part of the sound card), at regular intervals the ADC measures the microphone's analog signal and outputs a number representing the amplitude of the signal at that precise instant. This is called a “sample”. Before long there are a huge number of samples all arranged in chronological order—a kind of “spot-map” of the original waveform(Figure 6).

The sampling rate, sample rate, or sampling frequency defines the number of samples per second taken from a continuous signal to make a discrete signal. For time-domain signals, it can be measured in hertz (Hz). The three most sample rates are 11.025 kHz, 22.05 kHz and 44.1 kHz. The higher the sample rate is, the more samples that are taken and thus the better the quality of the digital audio.

A set of digital audio samples contains data that, when converted into an analog signal, provides the necessary information to reproduce the sound wave. The samples then code in the binary digit, it uses the bit depth. In digital audio, bit depth describes the number of bits of information recorded for each sample. Bit depth directly corresponds to the resolutions of each sample in a set of digital audio data. The two common bit depths are 8bits and 16bits, common examples of bit depth include CD audio, which is recorded at 16bits, and DVD-Audio, which can support up to 24 bits audio. The standard audio CD is said to have a data rate of 44.1kHz/16, implying the audio data is sampled 44,100 times per second, with a bit depth of 16. CD tracks are usually stereo, using a left and right track, so the amount of audio data per second is double that of mono, where only a single track is used. The bit rate is then 44100 samples/second*16bits/sample*2=1411 200bit/s or 1.4Mb/s.

The digital audio signal may then be stored or transmitted. Digital audio storage can be on a CD, a MP3 player, a hard drive, an USB flash drive, a compact flash, or any other digital data storage device. Audio data compression techniques—such as MP3, advanced audio coding, or Flac—are commonly employed to reduce the file size. Digital audio can be streamed to other devices.

The last step for digital audio is to be converted back to an analog signal with a digital-to-analog converter (DAC). Like ADCs, DACs run at a specific sampling rate and bit resolution but through the processes of oversampling, upsampling, and downsampling, this sampling rate may not be the same as the initial sampling rate, Figure 7 shows the process of digital-to-analog conversion.

After sample and code the digital audio signal may then be stored or transmitted. This audio data can then be stored uncompressed or compressed to reduce the file size. An audio file format is a container format for storing audio data on a computer system.

It is important to distinguish between a file format and a codec. A codec performs the encoding and decoding of the raw audio data while the data itself is stored in a file with a specific audio file format. Though most audio file formats support only one audio codec, a file format may support multiple codecs, as AVI does.

There are three major groups of audio file formats.

• Uncompressed audio formats, such as WAV, AIFF, and AU.

• Lossless compression formats, such as FLAC, Monkey's Audio (filename extension APE), WavPack (filename extension WV), Shorten, Tom's lossless Audio Kompressor (TAK), TTA, Apple Lossless and lossless Windows Media Audio (WMA).

• Lossy compression formats, such as MP3, Vorbis, Musepack, lossy Windows Media Audio (WMA) and AAC.

There is one major uncompressed audio format, PCM, which is usually stored as a .wav on Windows or as .aiff on Mac OS. WAV is a flexible file format designed to store more or less any combination of sampling rates or bit rates. This makes it an adequate file format for storing and archiving an original recording. A lossless compressed format would require more processing for the same time recorded, but would be more efficient in terms of space used. WAV, like any other uncompressed format, encodes all sounds, whether they are complex sounds or absolute silence, with the same number of bits per unit of time. As an example, a file containing a minute of playing by a symphonic orchestra would be the same size as a minute of absolute silence if they were both stored in WAV. If the files were encoded with a lossless compressed audio format, the first file would be marginally smaller, and the second file taking up almost no space at all. However, to encode the files to a lossless format would take significantly more time than encoding the files to the WAV format, Recently some new lossless formats have been developed (for example TAK), whose aim is to achieve very fast coding with good compression ratio.

Some of the common sound file format types are shown in the following.

• WAV—Standard audio file container format used mainly in Windows PCs. Commonly used for storing uncompressed (PCM), CD-quality sound files, which means that they can be large in size—around 10MB per minute. Wave files can also contain data encoded with a variety of codecs to reduce the file size (for example the GSM or MP3 codecs). Wav files use a RIFF structure.

• FLAC—A lossless compression codec. This format is a lossless compression as like zip but for audio. If you compress a PCM file to FLAC and then restore it again it will be a prefect copy of the original. (All the other codecs discussed here are lossy which means a small part of the quality is lost). The cost of this losslessness is that the compression ratio is not good. Flac is recommended for archiving PCM files where quality is important (e.g. broadcast or music use).

• AIFF (Audio Interchange File Format—The standard file format used by Apple. It is like a WAV file for the Mac.

• RAW—A RAW file can contain audio in any codec but is usually used with PCM audio data. It is rarely used except for technical tests.

• AU—The standard audio file format used by Sun, Unix and Java.

• VOX—The VOX format most commonly uses the dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. Similar to other ADPCM formats, it is compressed to 4-bit. Vox format files are similar to wave files except that the vox files contain no information about the file itself so the codec sample rate and number of channels must first to be specified in order to play a vox file.

• AAC—The Advanced Audio Coding format is based on the MPEG-2 and MPEG-4 standards. AAC files are usually ADTS or ADIF containers.

• MP3—The MPEG-1 audio layer 3 compression format is the most popular format for downloading and storing music. By eliminating portions of the audio file that are essentially inaudible, Mp3 files are compressed to roughly one-tenth the size of an equivalent PCM file while maintaining good audio quality.

• WMA—The popular Windows Media Audio format owned by Microsoft.

• RA—A Real Audio format designed for streaming audio over the Internet. The .ra format allows files to be stored in a self-contained fashion on a computer, with all of the audio data contained inside the file itself.

Today digital audio tends to be used more than analog audio as a method of storage. Records and cassette tapes continue to be used, but by a relatively small market. Why is digital audio more prevalent? While there is some debate as to whether or not digital audio actually sounds better than analog audio, digital audio is certainly easier to reproduce and to manipulate without loss of quality. Because of digital audio, it is much easier for both amateur and professional musicians today to produce studio-quality music.

New Words and Expressions

error n. 错误，误差

narration n. 叙述

waveform n. 波形

axis n. 轴，坐标轴

represent vt．表现，描绘，声称，扮演 vi． 提出异议

complicated adj． 复杂的，难解的

squiggly adv． 弯弯曲曲地

horizontal adj. 地平线的，水平的

amplitude n. 振幅，丰富，广阔

volume n. 音量，卷，册，体积

cycle n. 周期，循环 vi． 循环，轮转 vt． 使循环

frequency n. 频率，周期，发生次数

pitch n. 斜度，程度，倾斜 vt. 投，掷，定位于 vi．投掷，坠落，倾斜

converter n. 转换器

sample n. 采样，标本，样品 vt. 采样，取样，抽取……的样品，尝试

rate n. 比率，速度，等级，价格 vt. 估价，认定，鉴定等级 vi. 被评价

vibrate v.（使）振动，（使）摇摆

voltage n. 电压，伏特数

discrete adj. 离散的，不连续的

codec n. 编码解码器

uncompress vt． 未压缩

format n. 格式，形式，板式 vt. 格式化，安排……的格局

lossless adj． 无损的

lossy adj. 有损的

equivalent adj. 相等的，相当的，同意义的 n. 等价物，相等物

Exercises to the Text

1.Translate the following words and phrases into English.

（1）数模转换器（2）无损压缩（3）有损压缩（4）采样频率

2.Translate the following paragraphs into Chinese.

(1) Digital audio has emerged because of its usefulness in the recording, manipulation, mass-production, and distribution of sound. Modern distribution of music across Internet through on-line stores depends on digital recording and digital compression algorithms. Distribution of audio as data files rather than as physical objects has significantly reduced cost of distribution.

(2) A set of digital audio samples contains data that, when converted into an analog signal, provides the necessary information to reproduce the sound wave. The samples then code in the binary digit, it uses the bit depth. In digital audio, bit depth describes the number of bits of information recorded for each sample.

(3) After sample and code the digital audio signal may then be stored or transmitted. This audio data can then be stored uncompressed or compressed to reduce the file size. An audio file format is a container format for storing audio data on a computer system.

(4) The MPEG-1 audio layer 3 compression format is the most popular format for downloading and storing music. By eliminating portions of the audio file that are essentially inaudible, MP3 files are compressed to roughly one-tenth the size of an equivalent PCM file while maintaining good audio quality.

(5) Why is digital audio more prevalent? While there is some debate as to whether or not digital audio actually sounds better than analog audio, digital audio is certainly easier to reproduce and to manipulate without loss of quality. Because of digital audio, it is much easier for both amateur and professional musicians today to produce studio-quality music.