Audio sampling Concept

Digital audio systems reproduce the original sound by converting the acoustic waveform into a series of binary data. The device used in this step is A/D) it Samples sound waves at tens of thousands of times per second. Each sampling record the state of the original analog sound waves at a certain time point, which is called a sample. Connect a series of samples to describe a piece of acoustic waves. The number of samples per second is called the sampling frequency or sampling rate, in Hz ). The higher the sampling frequency, the higher the acoustic frequency that can be described. The sampling rate determines the sound frequency range (equivalent to the tone), which can be expressed in digital waveforms. The frequency range expressed in waveforms is generally referred to as bandwidth. Correct understanding of audio sampling can be divided into the number of sampling digits and the sampling frequency.

1. Number of sampling digits

The number of sampling digits can be understood as the resolution of the sound processed by the acquisition card. The larger the value, the higher the resolution, and the more real the recording and playback sounds. First, we need to know that the sound files in the computer are represented by numbers 0 and 1. Therefore, the essence of recording on a computer is to convert analog sound signals into digital signals. On the contrary, the digital signal is restored to analog sound signal output during playback. The bit of the acquisition card is the binary number of the digital sound signal used by the acquisition card to collect and play audio files. The position of the acquisition card objectively reflects the accuracy of the digital sound signal on the input sound signal description. 8 represents the Power 8 of 2-, and 16 represents the power 16 of 2-64 K. Compared with a piece of same music information, a 16-bit sound card can be divided into 64 K precision units for processing, while an 8-bit sound card can only process 256 precision units, this results in greater signal losses, and the final sampling effect is naturally incomparable.
Today, all mainstream products on the market are 16-bit acquisition cards, not the 64-bit or even 128-bit cards that some ignorant merchants advocate, they confused the concept of the secondary audio of the acquisition card with the concept of the number of sampling digits. Although the emu10k1 chip, which is currently the most powerful Collection Card series, claims to be 32-bit, it is only a multi-audio stream technology based on direct sound acceleration, in essence, it is a 16-bit sound card. It should be said that the 16-bit sampling accuracy is more than enough for computer multimedia audio.

2. sampling frequency

Digital audio systems reproduce the original sound by converting the acoustic waveform into a series of binary data. The device used in this step is A/D) it Samples sound waves at tens of thousands of times per second. Each sampling record the state of the original analog sound waves at a certain time point, which is called a sample. Connect a series of samples to describe a piece of acoustic waves. The number of samples per second is called the sampling frequency or sampling rate, in Hz ). The higher the sampling frequency, the higher the acoustic frequency that can be described.
Sampling frequency refers to the number of times the recording device samples the sound signal within one second. The higher the sampling frequency, the more real the sound is restored. In today's mainstream acquisition cards, the sampling frequency is generally divided into three levels: 22.05 kHz, 44.1 kHz, and 48 khz. 22.05 kHz can only achieve the sound quality of fmbroadcast, 44.1khz is the theoretical limit of CD sound quality, while 48 khz is more accurate. Human ears can no longer identify the sampling frequency higher than 48 khz, so there is little value for use on the computer.
The sampling rate of 5 kHz can only reach the sound quality of speech.
The sampling rate of 11 kHz is the lowest standard for playing a short video, which is 1/4 of the CD quality.
The sound at the 22 KHz sampling rate can reach half of the CD sound quality. At present, most websites use this sampling rate.
The sampling rate of 44 kHz is the standard CD sound quality, which can achieve good auditory effect.

3. Bit Rate

Bit Rate refers to the amount of information that can be passed per second in a data stream. You may have seen the situation described in the audio file "128-kbps MP3" or "64-kbps WMA. Kbps indicates "thousands of bits per second". Therefore, a larger value indicates more data:-kbps MP3 audio files contain twice the data volume of-kbps WMA files and occupy twice the space. (But in this case, the two files sound no different. Why? Some file formats can use data more effectively than other files. The sound quality of the 64-kbps WMA file is the same as that of the 128-kbps MP3 file .) It is important to know that the higher the bit rate, the larger the amount of information, the larger the processing capacity for decoding the information, and the more space the file occupies.
Selecting an appropriate bit rate for the project depends on the playback target: If you want to put the VCD to the DVD player for playback, the video must be 1150 kbps, and the audio must be 224 kbps. A typical 206 MHz Pocket PC-supported MPEG Video can reach 400 kbps-an exception occurs when the playback limit is exceeded.

VBR

 

VBR (variable bitrate) dynamic bit rate. That is, there is no fixed bit rate, and the compression software determines the bit rate based on the audio data during compression. This is an algorithm developed by Xing. They use high bitrate encoding for the complex part of a song and low bitrate encoding for the simple part. Although the idea is good, it is a pity that the Xing encoder's VBR algorithm is very poor, and the sound quality is far from that of CBR. Fortunately, lame perfectly optimizes the VBR algorithm to make it the best encoding mode for MP3. This is based on the quality of both the file size and the recommended encoding mode.

The average bit rate of ABR (average bitrate) is an interpolation parameter of VBR. Lame creates this encoding mode based on the poor file volume ratio of CBR and the variable file size generated by VBR. API is also called "Safe VBR". It is within the specified average bitrate and takes every 50 frames (30 frames about 1 second) as a segment, low-frequency and non-sensitive frequencies use relatively low traffic, and high traffic is used for high-frequency and high-dynamic performance. For example, when 192 kbps ABR is specified to encode a WAV file, Lame will fix the 85% encoding of the file with 15% kbps, and then dynamically optimize the remaining: the complex part is encoded at higher than 192 kbps, and the simple part is encoded at lower than kbps. And 192 kbps
Compared with CBR, the 192 kbps ABR file size is similar, but the sound quality is improved a lot. The speed of ABR encoding is two to three times that of VBR encoding. the quality in the range of 128-256kbps is better than that in CBR. It can be used as a compromise between VBR and CBR.

CBR (constant bitrate), a constant bit rate, indicates that a file is a bit rate from start to end. Compared with VBR and ABR, It compresses a large volume of files, but does not significantly improve the sound quality.
For MP3, bitrate is the most important factor. It is used to indicate the bit per second (BPS) occupied by the audio data ). The higher the value, the better the sound quality.

MP3

 

The full name of MP3 should be mpeg1 layer-3 audio files, and MPEG (Moving Picture Experts Group) should be translated as an expert group of active images in Chinese, especially for activity audio and video compression standards, an MPEG audio file is an audio part of the mpeg1 standard. It is also called an MPEG audio layer. It is divided into three layers based on the compression quality and encoding complexity, namely layer-1, Layer2, layer3, it corresponds to the MP1, MP2, and MP3 audio files respectively, and uses different levels of encoding according to different purposes. The higher the level of MPEG Audio Encoding, the more complicated the encoder and the higher the compression ratio. The compression ratios of MP1 and MP2 are and 6- 8respectively, while the compression ratio of MP3 is as high
-, That is to say, a minute of CD sound quality music requires 10 MB of storage space without compression, and only about 1 MB after being compressed by MP3. However, MP3 uses lossy compression for audio signals. To reduce sound distortion, MP3 adopts "sensory encoding technology", that is, Spectrum Analysis of audio files is performed before encoding, then filter out the noise level, and then sort the remaining bits in a quantitative manner to form an MP3 file with a high compression ratio, in addition, the compressed file can achieve sound effect close to the original audio source during playback. (In addition, the mp3pro: mp3pro encoder divides the audio recording into two parts: the MP3 part and the Pro part. Low
Frequency Band), and encode it into a normal MP3 file data stream. This enables the encoder to centrally encode less useful information and obtain better encoding results. At the same time, this ensures the compatibility between the mp3pro file and the old MP3 player. Pro analyzes the high frequency band information and encodes it into a part of the MP3 data stream, which is usually ignored in the old MP3 decoder. The new mp3pro decoder will effectively use this part of the data stream and combine the two segments (high-frequency and low-frequency segments) to produce a complete audio band to enhance the sound quality .)

 

-- From Baidu encyclopedia