Tag: what does audio format pcm mean

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Introduction to various conventional audio encodings (or formats) Part 3

Hearing model import: Experts have found that the human ear has a shadow effect through long-term acoustic research.

The sound signal is actually a type of energy wave, which propagates in air or other media. The most direct response of the human ear to the amount of sound energy, that is, the volume or pressure of the sound, is to hear the size of the sound. We call it the volume, which means the volume. The unit of energy is the decibel (dB). Even sounds of the same volume can be perceived by people as different in size due to their different frequencies. The 4000 Hz frequency is the easiest for the human ear to hear. It doesn’t matter if the frequency increases or decreases, even if the volume is the same, everyone will feel the sound becomes smaller. But when the volume drops to a certain level, the human ear cannot hear it, and each frequency has a different value.

You can see that this curve basically forms a V. When the frequency exceeds 15000 Hz, the human ear will feel that the sound is very small. Many people who are not very good at hearing cannot hear the frequency of 20000 Hz at all, no matter how loud it is… When the human ear hears two sounds with different frequencies and different volume at the same time, the one with the lower volume will also be ignored. For example, it is hard to hear the sound of the computer cooling fan during the day, but it becomes a noise source at night. According to this principle, the encoder can filter out many inaudible sounds to simplify information complexity and increase the compression ratio without significantly reducing sound quality. This shading is called the simultaneous shading effect. However, sound A is protected by sound B. If A is within the protection range centered on B, the protection will be more obvious. This range is called the critical bandwidth. The critical bandwidth of each frequency is different and the higher the frequency, the larger the critical bandwidth.

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Introduction to various conventional audio encodings (or formats)

1.PCM encoding

PCM Pulse Code Modulation is short for Pulse Code Modulation. In the text above, we mentioned the general PCM workflow. We don’t need to care which calculation method is used in the final PCM encoding. We just need to know the advantages and disadvantages of the PCM encoded audio stream. The biggest advantage of PCM encoding is good sound quality and the biggest disadvantage is its large size. Our common audio CD uses PCM encoding, and the capacity of one CD can only hold 72 minutes of music information.

2. WAVE

This is an old audio file format, developed by Microsoft. WAV is a file format that complies with the PIFF Resource Interchange File Format specification. All WAVs have a file header, the encoding parameters of this file header audio stream. WAV does not have strict rules for encoding audio streams. In addition to PCM, almost all encodings that support the ACM specification can encode WAV audio streams. Many friends do not have this concept. Let’s take AVI as an example, because AVI and WAV are very similar in file structure, but AVI has one more video stream. There are many types of AVIs we have come into contact with, so we often need to install some decoders to watch some AVIs. DivX, which we have come into contact with a lot, is a type of video encoding. AVI can use DivX encoding to compress video streams, of course, we can also use other code compression. Similarly, WAV can also use a variety of audio codecs to compress its audio stream, but we commonly use WAV whose audio stream is processed by PCM encoding, but this does not mean that WAV can only use PCM codec, it is also you can use MP3 codec. in WAV Just like AVI, as long as the corresponding Decode is installed, you can enjoy these WAVs.

On the Windows platform, WAV based on PCM encoding is the best supported audio format. All audio software can support it perfectly. Because it can meet higher sound quality requirements, WAV is also the preferred format for music creation and editing. Suitable for storing musical material. Therefore, WAV based on PCM encoding is used as an intermediate format, and is often used in the mutual conversion of other encodings, such as MP3 to WMA.

PCM Audio Coding Part 4

Bit rate

Bitrate refers to how many bits per second the encoded audio data must be represented.

lossy and lossless
For “lossless audio” we usually mean, it generally refers to the 16-bit/44.1kHz sample rate file format in the traditional CD format, and is called lossless compression because it includes the file format of 20Hz-22.05kHz. frequency response frequency that completely covers the audible range of the human ear.

Where I have confusion here is the relationship between the channel and the sample rate? At first, the sample rate was assumed to be 44100. If two channels were used, the sample rate of each channel would be 22100. This is actually incorrect, the sample rate is the sample rate of each channel, not the sample rate of all channels.
Therefore, if the sample rate is 44100, then for two channels the number of samples collected should be 88200.

PCM Audio Coding Part 3

Sampling frequency

The human frequency recognition range is 20HZ – 20000HZ. If 20,000 sound samples per second can be sampled, it can meet the needs of human ears during playback.

8000 Hz for telephone sampling.
A sample rate of 22050 is typically used.
44100 is already CD quality, and samples over 48000 are meaningless to the human ear
When decoding AAC (Advanced Audio Coding) audio with a sampling rate of 44.1 kHz, the decoding time of one frame should be controlled within 23.22 milliseconds. Usually it is a frame with 1024 sampling points.

Why do we have to talk about audio frames here?
The concept of audio frames is not as clear as that of video frames. Almost all video encoding formats can simply think of a frame as an encoded image. But the audio frame is related to the encoding format, which is implemented by each encoding standard. Because if it’s PCM (unencoded audio data), you don’t need the concept of frames at all, and it can be played according to the sample rate and sample precision. For example, for audio with a sample rate of 44.1 kHz and a sample precision of 16 bits, you can calculate that the bit rate is 44100 16 kbps and the audio data per second is fixed at 44100 16/8 bytes .
But we don’t want all the samples returned to us for processing, what we want is to return all the data sampled over a period of time. The audio box here is how much sample data is returned to us each time and in general 2048 sample data is returned.
So what is the size of 2048 sample data using 16-bit sampling bits for mono? 2048*16/8 = 4096 bytes.

Sampling bits
Each sampled data registers the amplitude, and the sampling precision depends on the size of the storage space (sampling bits):

1 byte (ie 8 bits) can only register 256 numbers, that is, only the amplitude can be divided into 256 levels
2 bytes (ie 16 bits) can be as small as 65536 numbers, which is already the CD standard;
4 bytes (i.e. 32 bits) can subdivide the amplitude into 4294967296 levels, which is really unnecessary
If it’s stereo, the samples are doubled and the file is almost doubled in size.

PCM Audio Coding Part 2

Coding

The quantized sampled signal is converted into a series of decimal digital code streams arranged according to the sampling sequence, ie, a decimal digital signal. A simple and efficient data system is a binary code system. Therefore, the decimal digital code must be converted to a binary code. Based on the total number of decimal digital codes, the number of binary code bits required can be determined, that is, the word length (sampling bits) The process of transforming the quantized sample signal into a binary code stream of a given word length is called encoding.

Example
Next, the above 1.65 V corresponds to a quantization level of 128. The corresponding binary system is 10000000. That is, the encoded result of the sample point is 10000000. Of course, this is an encoding method without considering values positive and negative, and there are many types of coding methods that require specific analysis of specific problems. (PCM audio format encoding is A-law 13 polyline encoding)

PCM audio encoding
PCM signals are not subject to any encoding or compression (lossless compression). Compared to analog signals, it is less susceptible to transmission system clutter and distortion. The dynamic range is wide and the sound quality is quite good. The coding adopts the A-law 13 polyline coding.

A-grade 13-fold line
The A-law is a form of logarithmic compansion in PCM non-uniform quantization. Digital pulse code modulation (PCM) is the basic method for digitizing analog signals today. PCM includes three steps: sampling, quantization, and encoding. Among them, the quantization is the discrete value of the sampling values. Uniform quantization and non-uniform quantization can effectively improve the quantization signal-to-noise ratio of the signal. Quantization of speech signals often takes two logarithmic forms of non-uniform quantization and ITU-recommended compression characteristics: A-law and Mu-law. A-law coding is primarily used in 30/32-channel group systems, and A- The PCM law is used in Europe and China.

See the article for more details

channel
Channels can be divided into PCM mono and stereo (two channels)
.Each sample value is contained in an integer i whose length is the minimum number of bytes needed to accommodate the specified sample length.
The low significant byte is stored first, and the bit representing the sample amplitude is placed in the high significant bit of i, and the remaining positions are 0, so the data format of the PCM waveform samples 8 and 16 bit is as follows.