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Reasons to use audio compression technology.<\/p>\n
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It is very easy to calculate the bit rate of a PCM audio stream, the value of the sample rate \u00d7 the value of the sample size \u00d7 the number of bps of the channel. A WAV file with a sample rate of 44.1 KHz, a sample size of 16 bits, and two-channel PCM encoding has a data rate of 44.1 K\u00d716\u00d72 = 1411.2 Kbps. We usually say that 128K MP3, the corresponding WAV parameter, is this 1411.2 Kbps, this parameter is also called data bandwidth, it is a concept with the bandwidth in ADSL. Divide the bit rate by 8 to get the data rate for this WAV, which is 176.4 KB\/s. This means storing a 1-second sample rate of 44.1 KHz, a 16-bit sample size, and a two-channel PCM-encoded audio signal, which requires 176.4 KB of space, which is approximately 10.34 M in 1 minute, which is unacceptable. For most users, especially friends who like to listen to music on the computer, to reduce disk usage, there are only 2 ways to downsample or compress. Lowering the index is not advisable, so experts have developed various compression schemes. Due to different uses and target markets, the sound quality and compression ratio achieved by various audio compression encodings are different, and we will mention them one by one in the following articles. One thing is for sure, they are all compressed.
\nFrequency vs. Sampling Rate
\nThe sample rate represents the number of times the original signal is sampled per second. The sample rate of most of the audio files that we see regularly is 44.1 KHz. What does this mean? Suppose we have 2 segments of sine wave signals, 20 Hz and 20 KHz respectively, each lasting one second, to correspond to the lowest and highest frequencies we can hear, and we sample these two signals at 40 KHz respectively. , we can get what kind of result? The result is: the 20 Hz signal is sampled 40K\/20=2000 times per vibration, while the 20K signal is only sampled 2 times per vibration. Obviously, under the same sample rate, the low-frequency information is much more detailed than the high-frequency information. This is also the reason why some audiophiles accuse CDs of digital sound not being real enough, and 44.1KHz CD sampling cannot guarantee that high-frequency signals are recorded well. To better record high-frequency signals, a higher sample rate seems to be required, so some folks use a 48KHz sample rate when capturing audio tracks from CDs, which is undesirable! Actually, this is not good for sound quality. For the ripping software, keeping the same sample rate as the 44.1 KHz provided by the CD is one of the guarantees for the best sound quality, rather than improving it. A higher sample rate is only useful for analog signals, if the signal being sampled is digital, do not try to increase the sample rate.<\/p>\n","protected":false},"excerpt":{"rendered":"