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What is the difference between 128k and 320k music?

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What is the difference between 128k and 320k music?

【Preface】

Some time ago, a colleague came across a very troubled client. The mess was said to have been caused by the client asking him to provide song files larger than 100MB-200MB in size. And my colleagues don’t know much about audio formats, so they started endlessly fumbling about FLAC, WAV and audio size. In the end, the colleague did not clearly explain to the customer what was going on.

After that, other things happened that made me feel that in the music industry there are too many practitioners around me who have an extremely poor understanding of music and even lack some basic knowledge related to music. I don’t even have the idea to understand, which makes me very sad. It seems that music has only one product attribute, and our practitioners only need to organize the shelves, encode various products, and recommend products to users with the big data of user purchase records, and they don’t need to worry about why the users users like this. Brands, what features these products have, and provide users with various services with cold data.

Therefore, I think it is necessary to write something. I don’t expect practitioners to become people who really love music. I just hope that even if you still think of “her” as a commodity, you can first figure out what you’re selling. and what is..

PS: The contents of the first lesson are multimedia files. Since the relevant content involves a lot of technical issues, it seems a bit boring, but if you read it carefully, you will see that it is actually very easy to understand, but these basic knowledge can be very helpful.Improve your skill well. Also expect more interesting content about records, musical styles, etc. which I will post soon.

Bit Rate, Sample Rate, Lossless, MP3, FLAC, APE, 320kb, 192kb, 128kb, 44.1khz, CBR, VBR. Does this bunch of various names make you both familiar and unknown?

The higher the bitrate, the better the sound quality. Lossless music is the highest sound quality, right? So, let’s start with the sound collection.

【Audio composition】

Nowadays, when we talk about audio, everything is digital audio. Digital audio consists of three parts: sample rate, sample precision, and number of sound channels.

Sample Rate: Both the sample rate, which refers to the number of samples per second when recording the sound, expressed in Hertz (Hz).

Sampling Precision: Refers to the dynamic range of the recorded sound, measured in bits (Bit).

Sound channel: the number of channels (1-8).

In simple terms, we can think of a sound wave as a curve. We know that the curve is made up of points, and the sampling rate is the number of points in the middle of the length per second (the horizontal axis in the figure above). Sampling precision is the number of points in the dynamic range (upper vertical axis). The finer the positioning of these two dimensions, the greater the true sound restoration and the better the sound quality. Of course, the larger the audio file will be.

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What is the proper audio bitrate? Part 2

What does audio bitrate mean?

Are higher audio bitrates better?

Encoding rate (Kbps) * total length of the song (seconds) / 8 = file size (KB), if divided by 1024, it is the size in MB.
mp3 is lossy compression, the smaller the file, the higher the loss. The relationship between bitrate and audio and video compression is simply that the higher the bitrate, the better the audio quality and video, but larger is the encoded file; yes The opposite is true for lower bitrates.
Bitrate refers to the sampling rate at which digital sound is converted from analog to digital format. The higher the sampling rate, the better the quality of the restored sound.
The bitrate value is compared to the actual audio:
16 Kbps = phone sound quality
24 Kbps = increase phone sound quality, short wave transmission, long wave transmission, European standard medium wave transmission
40 Kbps = American standard medium wave transmission 56 Kbps
= voice
64 Kbps = voice boost (best bit for mobile phone ringtones) = tape (best setting for mobile phone stereo MP3 player, best setting for low-end MP3 player
112 Kbps = FM stereo radio FM 128 Kbps ) 160 Kbps = Hi-fi HIFI (best setting for mid-range MP3 player to high-end MP3 players) 192 Kbps=CD (best setting for high-end MP3 player) high-end ) 256 Kbps= Studio Music Studio (for music enthusiasts) In fact, with the advancement of technology, bit rates are also getting higher and higher, MP3 has a maximum bit rate of 320 Kbps, but some formats can achieve higher bit rates and superior sound quality. For example, the emerging APE audio format can provide true audiophile-level lossless sound quality and smaller volume than WAV format, and its bit rate is typically 550kbps.

What is the proper audio bitrate?

What does audio bitrate mean?

What is the proper audio bitrate? When recording gameplay videos and putting them on the Internet, the bit rate used to suppress the video depends on the end use. If you are going to share it on an online video site like Youku, the requirements for videos will vary from site to site. Youku’s requirements for video formats are as follows: Generally speaking, when uploading videos to online sites, it is recommended that the video be encoded in H264/X264, and the video bitrate should be 1600Kbps;
What is the proper audio bitrate?

When recording gameplay videos and putting them on the Internet, the bit rate used to suppress the video depends on the end use.

If you are going to share it on an online video site like Youku, the requirements for videos will vary from site to site. Youku’s requirements for video formats are as follows:

Generally speaking, when uploading to online video website, it is recommended that the video be encoded by H264/X264, the video bit rate should be 1600Kbps; audio must be AAC encoded and audio bitrate must be 128 Kbps.

If you plan to put it on the network drive and share it with friends, it is recommended to use a higher bit rate. For example, the video is encoded by H264/X264 and the video bit rate is 2400Kbps; the audio is AAC encoded and the audio bitrate is 128 Kbps.

What does audio bitrate mean?

The code rate is the number of data bits transmitted per unit of time during data transmission. Generally, the unit we use is kbps, that is, kilobits per second.
A common understanding is the sample rate. The higher the sample rate per time unit, the higher the precision, and the processed file will be closer to the original file, but the file size is proportional to the sample rate, so almost all encoding formats pay attention to This is how to use the lowest code rate to achieve the least distortion. The cbr (fixed code rate) and vbr (variable code rate) derived from this core are all items in this regard, but things are not absolute, In terms of audio, the higher the bit rate, the lower the compressed ratio, the smaller the sound quality loss and the closer it is to the sound quality of the audio source.
Basically, the sound quality of the two data 44.1 and 128 in the MP3 song attribute is very good.

What do the bits, bit rate and sample rate of an audio file mean? Part 2

If it’s in a lossless uncompressed format, the bit rate is strictly equal to the number of bits * sample rate * number of channels. And typically, the MP3 bitrate you can see just represents how much capacity the format needs to describe this one second of audio.

MP3 is a lossy compression. In the compression process, some information is lost, but the lost information cannot be represented by the number of bits and the sampling rate. In general, the higher the bit rate, the less information will be lost. Mathematically, bitrate and sound quality are proportional. As for whether you can hear it or not, it depends on many factors. The MP3 algorithm is not complicated, of course, to understand it you have to learn what the Fourier transform is.

There is also lossless compression (representing APE, FLAC, etc.), which also has a bitrate, and this bitrate has nothing to do with sound quality. It also describes how much capacity the file uses to describe one second of audio content, but the same audio content can be compressed to different sizes (compression ratios), similar to zip compression ratios. No matter how big you compress it, in the end it can be restored to the same file. So if you see who is chasing lossless bitrate, you can basically conclude that the product is a bad pen.

What do the bits, bit rate and sample rate of an audio file mean?

For example, the common mp3 format audio source is 16-bit 320 Kbps or Sony Hi-Res 24-bit 192 KHz.

The stored sound data (for uncompressed PCM format) is equivalent to drawing a curve on graph paper, and its points can only be on the grid.

Then the sampling rate is how dense the grid is in the horizontal direction (how many grids per second); the number of digits is how dense the grid is in the vertical direction (how thin and how much storage each sample is stored), 16 Bit means that the range of values of each sample point is from 2 to the 16th, i.e. , from 0 to 65535. The code rate is the multiplication of the two (how much storage the samples take per second).

First, there are two cases, lossy and lossless. After figuring out lossless and lossless, we need to figure out the sample rate and bit depth. After calculating the sample rate and bit depth, we need to figure out the difference between the ratio of bit rate and bit depth to sample rate and its impact on audio quality.

In order to store a continuous physical signal (well, tell me about Planck’s constant…) in a computer, it must be converted to a digital signal. In acoustics, a digital signal is a digital representation of the amplitude of the sound wave at any moment.

Sound waves are longitudinal waves, which are difficult to draw. The following figure is replaced by transverse waves (the concept of longitudinal waves is the phenomenon that the density of air or other media changes regularly due to energy. The peaks represent high density, the troughs represent low density, and the horizontal line is the average density, i.e. silent state)

Using high school physics, waves contain two dimensions, one is intensity and the other is time. “Number of digits” indicates how many levels the sound wave is divided into from the strongest to the weakest; “Sampling Rate” determines the precision of the time axis or the sampling density, i.e. the length of time represented by each red dot, and the code rate is one second The number of dots on the clock, multiplied by the space that each point occupies.

So the so-called 24 bits consist of dividing the intensity of the sound wave by 2 at power level 24, occupying 3 bytes of space. Obviously, the finer the grade, the more details are restored.

The sample rate is generally 44100 Hz for CD (Hertz = times/second), 48000 Hz for DVD, and 96000 Hz as standard. As with the number of digits, the more points you get in a single second, the more details are restored. Why does CD take this value? Because the hearing range of the human ear is generally believed to be between 20 and 20,000 Hz. It is necessary to represent a peak and a valley, and at least two sampling points are required. So the CD can represent the sound of 22050 Hz at most, but this sound doesn’t have any detail, because if there are only two peak and trough points, the mean waveform is completely lost. Therefore, there will be a higher sampling rate.

Audio bit depth, sample rate, and bit rate

About the sample rate. What is the most intuitive effect of sample rate?

Affects the expressiveness of the sound’s frequency range. The higher the sample rate, the larger the frequency range that can be expressed. 44.1KHz sampling rate can represent the frequency range from 0Hz to 22050Hz; 48KHz sampling rate can represent the frequency range from 0Hz to 24000Hz; 96 KHz sample rate can represent the frequency range from 0 Hz to 48000 Hz. The average frequency range that the human ear can hear is approximately 20 Hz-20,000 Hz. Combining the above two, if you see one parameter: 16Bit 44.1KHz , it means this digital audio can express “96dB dynamic range” and “0 Hz -22050 Hz” frequency range; 24Bit 48KHz , it means this digital audio can Performance ” 144dB Dynamic Range ” and ” 0Hz -24000 ”

Hertz” frequency range. (3) Audio bit rate, also called bit rate, or bit rate. Bit rate refers to the amount of information that can pass through a data stream per second, which can also be understood as: how much is used per second The amount of data in bits to represent In principle, the higher the audio bitrate, the better the quality However, if it is lossy compressed audio , different compression algorithms, even if the bitrate is the same, will lead to completely different sound quality results.Typical representative: 96kbpsThe sound quality of WMA audio format is obviously better than that of MP3 audio format than 96 kbps. Why is this? The difference is due to different compression algorithms and different data utilization rates. Another example, if the MP3 is compressed below 48 kbps, it’s already terrible. And if it’s format AAC audio, to the same 48kbps bit rate, the sound quality is obviously better than MP3. For lossless compressed audio, even if the bitrate is completely different, the final sound quality is the same

. For example, if the same WAV file is compressed in FLAC and APE formats, the bit rate of the output file is not the same, but the sound quality is the same. Even if it is the same format, the compression level is different and the bitrate is completely different, but the end result, the sound quality remains the same (but when encoding and decoding, the CPU usage is different and the encoding time is also different).

MP3 is just an audio format.

MP3 Quality

And the audio has several important parameters, such as KHZ, BIT, channel, KBPS, etc.

MP3 Quality

And the format is different, the algorithm is also different, so even if the above parameters are the same, the format is different. The sound quality will also be very different. Among them, VBR is a dynamic sampling. For a detailed and complete explanation, please refer to the description below.
After reading it patiently, you will be able to say a thing or two.

Audio sampling explains the digital audio system The original sound is reproduced by converting the waveform of the sound wave into a series of binary data The equipment used in this step is an analog-to-digital (A/D) converter, which samples the sound wave at a rate of tens of thousands of times per second. A sampler records the state of the original analog sound wave at a given time, called a sampler. A series of samples can be connected to describe a sound wave. The number of samples per second is called the sample rate or sample rate, and the unit is HZ.

(hertz). The higher the sample rate, the higher the frequency of the sound wave that can be described. The sample rate determines the frequency range (equivalent to the pitch) of the sound, which can be represented by a digital waveform. The frequency range is often called the bandwidth. To understand properly, audio sampling can be broken down into the number of bits sampled and the frequency of the sample. 1. The number of sampling bits The number of sampling bits can be understood as the resolution of the sound processed by the capture card. The higher the value, the higher the resolution and the more realistic the sound recorded and played back. The first thing we need to know: sound files on the computer are represented by the numbers 0 and 1 . So the essence of recording on the computer is to convert the analog sound signal into a digital signal. On the contrary, during playback, the digital signal is restored to an analog sound signal output. The capture card bit refers to the binary digits of the digital sound signal used by the capture card when capturing and playing sound files. The bits on the capture card objectively reflect the accuracy of the digital sound signal’s description of the input sound signal. 8 bits represent the eighth power of 2–256 and 16 bits represent the sixteenth power of 2–64K . For comparison, for the same music data, a 16-bit sound card can break it down into 64K precision units for processing, while an 8-bit sound card can only handle 256 precision units.

It causes a large signal loss, and the final sampling effect is naturally incomparable. All the major products on the market today are 16-bit capture cards, not 64-bit or even 128-bit as some ignorant marketers advocate, who confuse the concept of polyphony in capture cards with the concept of bits of sampling. Although the EMU10K1 chip used by the most powerful capture card series is claimed to be 32-bit, it is just a Direct Sound acceleration-based multi-audio streaming technology, and its essence is still a 16-bit sound card. . It must be said that the 16-bit sampling precision is more than enough for computer multimedia audio. 2. Audio Sampling Level (Audio Sampling Rate) The digital audio system reproduces the original sound by converting the sound waveform into a series of binary data. The equipment used to accomplish this step is an analog-to-digital (A/D) converter, which takes each sound wave, and each sampler records the state of the original analog sound wave at a given time, which is called sample. A series of samples can be connected to describe a sound wave. The number of samples per second is called the sample rate or sample rate, and the unit is HZ (Hertz). The higher the sample rate, the higher the frequency of the sound wave that can be described.

The higher the sample rate, the better?

First of all, we need to understand what sampling rate is. Sample rate refers to how many sound samples the computer collects per second. Generally speaking, the higher the sampling rate, that is, the shorter the sampling interval, the more sound sample data the computer can obtain per unit of time, and the more accurate the sound waveform representation will be. sound.

So the higher the audio sample rate of the MP3 chip, the better? In principle: the higher the sample rate, the better the sound quality.

Sampling frequency is generally divided into three levels: 22.05KHz, 44.1KHz and 48KHz; 22.05 KHz can only reach the sound quality of FM radio, 44.1 KHz is the theoretical limit of CD sound quality, and 48 KHz has reached DVD sound quality. The low and high frequencies that our human ear can perceive are 20 Hz and 20 KHz, so to meet the hearing requirements of the human ear, at least 40,000 samples per second are required.

expand:

8000 Hz: the sampling rate used by phones, sufficient for human speech

11,025 Hz: the sample rate used for AM broadcasts

22,050 Hz and 24,000 Hz: sample rates used by FM radio

32,000 Hz: Sampling frequency used by miniDV digital video camera, DAT (LP mode)

44,100 Hz: Audio CD, also sample rate commonly used for MPEG-1 audio (VCD, SVCD, MP3)

47,250 Hz: the sample rate used by commercial PCM recorders

48,000 Hz: sample rate used for digital sound used in miniDV, digital TV, DVD, DAT, movies, and professional audio

50,000 Hz: the sample rate used by commercial digital recorders

96,000 or 192,000 Hz: Sample rate used by DVD-Audio, some LPCM DVD audio tracks, BD-ROM (Blu-ray Disc) audio tracks, and HD-DVD (high-definition DVD) audio tracks. )

2.8224 MHz: The sample rate used by Direct Stream Digital’s 1-bit sigma-delta modulation process

What does the quality of the audio or video depend on?

What factors are crucial for a higher or lower quality in digital files?

Files like WAV are not compressed, but they take up a lot of space and contain a lot of inaudible and unnecessary information.

This forced them to look for ways to be able to compress this audio or video to save space and so that they were also lighter to watch or listen to live on sites like YouTube or Spotify, etc.

Let’s see what are the important parameters:

1.- Bit Rate

It is the bit transfer rate. Said in an easy to understand way, it is the amount of information that can be transmitted per second. It is obvious that the more information that can be transmitted, the greater the detail and therefore the greater the quality.

Sometimes the transfer rate is fixed (CBR) but it can also be variable. Where the bitrate chosen for each moment will depend on the amount of information needed.

An audio with a lot of silence and few sounds will need to transfer much less information per second than an audio full of sounds and nuances.

2.- Sampling rate

The amount of information per second that is sampled will mark the number of details that are captured and can be transmitted.

But obviously also the higher the sampling rate, the greater the information contained.

Therefore, we see as a very simplified summary that a higher quality requires greater detail and therefore better information stored and a higher speed of transmission of that information.

Therefore higher bitrate and higher samplerate will usually mean higher quality, and will also take up more space.

Which is better, MP3, FLAC or WAV? Part 2

The right audio file format can do more with less

M4A

M4A is a format compatible with Apple products such as iPhone and iPod, and corresponds to MPEG-4 video files. M4A files have lossless compression and can be edited multiple times without quality loss. Compared to other audio file formats, M4A has similar quality but smaller file size.

CCA

Like MP3, Advanced Audio Coding (AAC) files are also lossy formats. However, AAC files are of much higher quality in their original state than all other audio file formats, and AAC files are generally similar in size to MP3, but slightly higher in quality.

In addition, AAC also offers variable bitrate or constant bitrate encoding, and AAC files are also open source, which means you don’t have to pay royalties to create and share them (unlike MP3 files). .

AAC files are usually used in iTunes, but can also be used in other player devices and game consoles.

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OGA

Like AAC, OGA files are open source and use lossy compression. They use a variable bit rate to reduce file size without reducing the overall sound quality. In practice, OGA files tend to be of higher quality than MP3s, but many programs don’t support OGA.

FLAC

Like M4A, FLAC is a lossless format, but unlike M4A, FLAC is an open source encoding and has a more efficient compression algorithm that reduces the file to 50% to 70% of its original size. This format is popular with audiophiles as a way to collect music in its highest quality form. Because it is open source, it is compatible with many devices and programs.

PCM, WAV and AIFF

PCM (Pulse-Code Modulation) is an uncompressed audio file format commonly used on CDs and DVDs. This format uses a “sample rate” and a “bit depth”, where the sample rate represents the sample rate of the original audio file and the bit depth represents how many bits are used to define each sample.

WAV (Waveform Audio File Format) is generally used as a format for storing PCM audio files, which makes it more suitable for Windows computers.

AIFF (Audio Interchange File Format, Audio Interchange File Format) like WAV is commonly used to store audio files in PCM format (except Mac).

How to choose the correct audio file format

Choosing the right format completely depends on the audio file you intend to use, you should choose a format that expresses the sound quality of the desired audio file, unnecessary high quality audio files can be difficult to move, share, convert and manage . Consider the following two cases:

If you plan to edit audio files (such as podcasts), use uncompressed formats (including PCM, WAV, and AIFF) so that the quality of the audio files is maintained each time you edit and save the file. Once completed, it can be generated in a compressed format that is easy to share.
If you are an audiophile and want to listen to your music in high fidelity, choose a lossless compressed file format, including FLAC, WMA, M4A. It uses less storage space than uncompressed formats, but still retains the original full file quality.
If you don’t care about the quality of audio files, want to share files on the Internet, or need to save hard disk space, choose a format that uses lossy audio file compression, including MP3, OGA, AAC. Fortunately, lossy compression has gotten so good in recent years that most people can’t tell the difference between lossy and lossless compression.
In conclusion

In summary, the most popular types of audio files are:

MP3
M4A
CAA
OGA
FLAC
WAV
PCM
AIFF
In some cases it is better to use one file format than the other, in which case you should convert between the two file formats. To programmatically convert between audio or video file types, you can use Filestack’s Audio Transcoding API or Video Transcoding API. The Filestack API also supports splitting audio files from video files, cropping or resizing video files, video file thumbnails, and more.