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The ratio of frequency, bit rate, bits rate and MP3 sound quality Part 4

Compared to 192, the compression rate of 256 kbps is naturally a higher step in sound quality.

Take the first 10 seconds of the track, the low frequency of the cello is obviously less granulated and the sound is softer and more natural. The texture is also clearer, the details are much more, the representation of the atmosphere is more prominent, the rotation of parts in the following songs is also more expressive, the clarity of the large and small signals, and the sound is also improved It is more careful and audible, but at the same time, the file size also increases at 6831 KB at a time, which is still affordable for an MP3 player of 256 m. It is not difficult to calculate that it can store about 135 minutes of music according to the Bit rate of 256. In general terms, it is enough, 128 m is a little less and can only support a little more than an hour, so it is recommended to use a Bit rate of 192 for 128 m.

320 Kbps is the maximum bit rate that LAME can provide. The final generated file is 8592 KB, which is approximately 8.4 M. Compared to 37 m from the WAV file, the compression ratio is basically 4.5: 1, but the generated MP3 file sounds very distorted compared to others Bit rates, the natural advantage of 320 is obvious. The bell, the details, etc. They are very delicate. Basically it achieves the sound quality of the original copy of the CD, especially in the CD player with the MP3 playback function. There is no difference, but I used earplugs relatively high-end ears with high resolution, in addition to my experience and skill with music and team, I can still hear much difference compared to the WAV file, first, the sounds MP3 tablets. a little different. The sensation of contraction is relatively dry in general. Without the WAV file, it sounds fresh and dynamic. In terms of final details, nuances and sense of space, the degree of separation is not as high as the WAV, but it is quite similar in terms of a ringer, it is a little less expressive and the digital flavor is relatively strong. Then, if you are using a miniature hard drive as an iPod, I recommend that you use a compression ratio of 320 kbps, you can get the best auditory experience. Of course, listen to WAV directly is the best ~ ~ without compression, without loss, but unfortunately there is no Walkman who admits compression without loss as APE, otherwise, there will be a variety of options.

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The ratio between frequency, bit rate, bit rate and sound quality of MP3 Part 3

First I took the track with RAC, and then I used the MP3 LAME encoder (VISION 1.92 Motor 3.92) engine in the CD’EX software to process the WAV file. I tried the parameters lick one by one to choose a good effect:

The first thread priority parameter selects the highest and the lowest respectively. When other parameters are equal, the comparison is compressed. It is found that the degree of thread priority has no effect on sound. The size of the files generated is the same. , and the comparison sounds the same, so this parameter has no effect on sound quality.

The second parameter is the version, which can be selected from MPEGI, MPEGII and MPEGII.V. The same, other parameters are determined and these three options are used to compress three times. The size is the same, but MPEGI’s real listening direction is better. The medium-low frequency compression ratio is a bit smaller, but high frequency distortion is a bit more. It is more suitable for listening to human voice and pop music. For classical music, use the MPEGI type. It is not bad, the sound is better, but if it is solo music with many medium and high frequencies such as the violin, it is recommended to use the MPEGII.V type, which will have better results.

The third parameter is the most important, which is the bit rate. Choosing it directly affects the size and auditory experience of its MP3 file. The greater the compression ratio, the greater the distortion, and the lower the compression ratio, the minor is distortion, but how can we find a us? What is the balance that is acceptable for both parties? This requires careful exploration in the experiment. Taking into account that the sound quality of the files with a low bit rate is not suitable for playing music, the minimum assembly is 128 kbps, and four fixed bit rate of 128, 192, 256 and 320 are used For comparison and test.

The compression ratio of 128 kbps remains relatively approximate, and the high frequency part is very distorted after compression. It sounds hollow, wrinkled, rough and, often, blinking. Mally understood, the compressed volume of a musical piece at 3 minutes 39 is 3414 kb. Although the volume is not great, the sound is not pleasant and there is a great flaw.

The compression effect of the Bits rate of 192 Kbps is much better than that of 128. First, the sound is solid, at least there is no vacuum sensation, high frequency distortion is also much lower, sound is compact, the noise is small and clean, which is ideal for contrast. The listening effect is only because the compression is still relatively strong, the performance of the details is not very good, the texture of musical instruments, especially the Wind instruments, it is still very hard. , unreal and lacks musicality. The compressed size is 5123 KB, I think this compression ratio is better to use in an MP3 player with a capacity of 128 ~ 256 m, which can not only meet the main sense of hearing, but is also suitable for size 128m You can store about 95 minutes of music, and 256m can duplicate 190 minutes of music.

The relationship between frequency, bit rate, bit rate and MP3 sound quality Part 2

Finally, if you want sound quality completely without loss, you should still use audio files in a lossless compression format or an uncompressed file format.

How good is the sound quality in mp3 format? 128/192/256/320, etc. What is the difference between the sound quality MP3 of various compression relationships / compression modes? What are some basic principles? How about the sound quality of other formats such as Ape / WMA / etc?

Speaking of MP3, I think I’m afraid that nobody will say that he has never heard about him. Even if you are not an User MP3, advertisements, advertising activities, discussions between friends and online resources, always give a small impression, right? For fashionable young people, especially friends who like music and friends who like digital devices, MP3 is probably a word from which you should talk every day, but what exactly mp3 , How to determine the sound quality of MP3 and how does it work? Is it good or bad? How can I hear high quality MP3? ? ? I think the following article can help you solve many doubts.

Through today’s MP3 users, the standard generally accepted for production is EAC capture + compression LAME. I also use this combination. Friends who have experience in this production process will discover some tricks for different music uses different configurations of parameters and compression ratios. ranging from 128 Kbps standard up to 320 Kbps maximum, but what is the effect and difference between these bit rates? ? How is the most appropriate compression ratio, what should be better for CBR and VBR, etc. These issues are often discussed by all. To clarify these details, I did a useful experiment and share it with you below. You will have some feelings.

Usually, I really like to listen to classical music, so I chose the first track of Bach’s “Grandenburg Concert” for this test. The version number of the FOOB2000 V0.8 playback software and the listening headphones are ER6 of Intech and E3C from Shure. . Because the classic repertoire has many details and the band is great, the requirements for all aspects of sound quality are relatively high, so it may clearly reflect the difference of details between the different processing methods.

The relationship between MP3 frequency, bit rate, bit rate and sound quality

I want to know the relationship between MP3 frequency, bitrate, bitrate and sound quality.

The higher the frequency and bit rate, the better the sound quality. It seems that most MP3 frequencies are 44100HZ. The bit rate is 128, 192, and so on.

The frequency mentioned here is the sampling rate, which is usually 44100 KHz, because this is the standard for music CDs.

Each song is ripped from a CD, converted to a WAV file, and then converted to MP3 using software like Lame. So it should be a sample rate of 44100 KHz. Unless yours isn’t a song, it’s recorded as a WAV file and you’ve selected other sample rates when recording.

The main factor that affects the sound quality of MP3 is the bit rate. The best today are 320K CBR (fixed bit rate) and VBR (variable bit rate), VBR files are a bit smaller than CBR. 192K VBR is the most popular on the Internet, which can meet the requirements of sound quality and file size at the same time, but I usually use CD to rip tracks or download APE (lossless compression, can be restored to WAV file) and then convert it to 320K VBR.

One last reminder: MP3 transcoding is distorted and the distortion cannot be reversed. In other words, if you convert MP3 to WAV sound quality, the file size increases dozen times, but the sound quality remains the same as MP3 sound quality.

If you want to hear low distortion, it’s better to listen to a CD or download APE.

First of all, sound quality is a very subjective thing!

It is often said that the sound quality is good, one refers to the good degree of reproduction, that is, the smaller the difference with the recording, the better; As for mp3, mp3 is a compressed format, the higher the bitrate, the less compression and less loss of detail, that is, the higher the bitrate, the closer to the original sound. But sound quality is also related to your output device. For example, a good mp3 player and a good pair of headphones will help your listening quality.

So if you want to improve sound quality, it’s best to start from the above perspectives and not overemphasize any one of them. When you have higher requirements for sound quality, you can give up mp3 and directly switch to stop CD. The CD carries wave files, which are completely lossless sound quality formats, which will perform better.

Assuming the distortion is small, the only way is to increase the bitrate. It’s best to use variable bit rate (VBR) compression to produce mp3 files, which can strike a balance between maximum fidelity and minimum file size.

Sample rate and bit rate of MP3 Part 2

The sampling process consists of extracting the frequency value of a certain point.

Obviously, the more points that are extracted in one second, the richer the frequency information that can be obtained. To restore the waveform, there must be two sampling points in one vibration. The highest frequency that can be felt is 20kHz, so to meet the hearing requirements of the human ear, at least 40k samples per second are required, expressed in 40kHz, and these 40kHz are the sampling frequency. Our common CD has a sample rate of 44.1 kHz. It is not enough to have only frequency information, we must also obtain and quantify the energy value of this frequency to represent the strength of the signal. The number of quantization levels is an integer power of 2, and the sample size of our common CD bit is 16 bits, that is, 2 to the power of 16. Sample size is harder to understand than bit rate. sampling, because it makes it seem abstract. For a simple example: suppose a wave is sampled 8 times, and the energy values ​​corresponding to the sampling points are A1-A8, but we only use 2-bit sampling size, as a result we can only keep the 4 point values ​​in A1-A8 and discard the other 4. If we use the 3bit sample size, all 8 point information is recorded. The higher the sample rate and sample size values, the closer the recorded waveform is to the original signal.

It is very easy to calculate the bit rate of a PCM audio stream, the value of the sample rate × the value of the sample size × 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×16×2 = 1411.2 Kb/s. We usually say that 128K MP3, the corresponding WAV parameter, is this 1411.2Kb/s, this parameter is also called data bandwidth, it is a concept with the bandwidth in ADSL. Divide the code rate by 8 to get the data rate for this WAV, which is 176.4 KByte/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.

The minimum value of the 16-bit binary number is 0000000000000000, the maximum value is 1111111111111111, and the corresponding decimal numbers are 0 and 65535, that is, the difference between the maximum and minimum values ​​is 65535, that is, the dynamic range of the analog quantity that quantizes The difference can be 65535, which is 96.32 decibels, so quantization accuracy is only related to dynamic range and has nothing to do with frequency response. It also makes sense to set the dynamic range to 96 decibels. The painless limit sound pressure of the human ear is 90 decibels. The dynamic range of 96 decibels is sufficient for ordinary applications. Therefore, after quantization, analog waves within the 96 decibel dynamic range will not be quantized. Clipping distortion will occur.

The number of digits in the sound is equivalent to the number of colors on the screen, indicating the amount of data per sample. Of course, the larger the amount of data, the more accurate the playback sound, so as not to confuse the sound. of the teapot with the train whistle. In the same way, it is more clear and precise for the image, so as not to confuse blood and ketchup. However, limited by the function of human organs, 16-bit sound and 24-bit image are basically the limit of ordinary human beings, and the higher digits can only be distinguished by instruments. For example, the phone has 7-bit sound sampled at 3 kHz and the CD has 16-bit sound sampled at 44.1 kHz, so the CD is clearer than the phone.

All major products on the market today are 16-bit capture cards, not 64-bit or even 128-bit as advocated by some ignorant marketers, who confuse the polyphony concept of capture cards with the concept of sampling bits.

MP3 sample rate and bit rate

When we listen to mp3 and watch movies, we will notice two parameters. The most common ones are 44.1 KHz sample rate and 192 Kbps bit rate. So what is the sample rate and what is the bit rate? What is the relationship between them?

Explain:

The process of converting an analog audio signal to a digital audio signal is called sampling. In a nutshell, how many data points does it take to record a 1 second long sound via waveform sampling. For example: the sound sample rate of 44.1 KHz is equivalent to spending 44,000 data points to describe the sound waveform for 1 second. 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; 22.05KHz can only achieve FM radio sound quality, and 44.1KHz is the theoretical limit of CD sound quality, 48KHz has reached DVD quality.

Sampling rate refers to the sampling frequency when converting sound (analog signal) to mp3 (digital signal), i.e. how many data points are sampled per unit of time. (The data for a sample point is 8 (or even more) bits long.)

Bit rate refers to the number of bits (bits) transmitted per second. The unit is bps (bit per second). The higher the bitrate, the more data transmitted and the better the sound quality.

It can be said that the sample rate and bit rate are like the horizontal and vertical coordinates on the coordinate axis. The sampling frequency on the abscissa represents the data points sampled per second. The bit rate of the ordinate represents the precision when quantizing analog quantities with digital quantities.

The sample rate is similar to the number of frames of moving images. For example, the sampling rate of movies is 24 Hz, the sampling rate of PAL format is 25 Hz, and the sampling rate of NTSC format is 30 Hz. When we play back the still images sampled at the same rate as the sampling frequency, we see a continuous image. In the same way, when a CD recorded at a sampling rate of 44.1 kHz is played back at the same rate, a continuous sound can be heard. Obviously, the higher the sample rate, the more coherent the sound will be heard and the picture will be seen. [Of course, the sampling rate that human auditory and visual organs can distinguish is limited, which is basically higher than sound sampled at 44.1kHZ, and most people haven’t noticed the difference. ]

The number of digits in the sound is equivalent to the number of colors on the screen, indicating the amount of data per sample. Of course, the larger the amount of data, the more accurate the playback sound, so as not to confuse the sound. of the teapot with the train whistle. In the same way, it is more clear and precise for the image, so as not to confuse blood and ketchup. [However, limited by the function of human organs, 16-bit sound and 24-bit image are basically the limits of ordinary humans, and the higher digits can only be distinguished by instruments. For example, the phone has 7-bit sound sampled at 3 kHz and the CD has 16-bit sound sampled at 44.1 kHz, so the CD is clearer than the phone. ]

When you understand the above two concepts, bitrate is easy to understand. Take the phone as an example, 3000 samples per second, each sample is 7 bits, then the phone’s bit rate is 21000. And the CD is 44100 samples per second, two channels, each sample is 13 bit PCM encoded, so the CD bit rate is 44100*2*13=1146600, which means the CD data volume per second is about 144KB. the capacity of a CD is 74 minutes equal to 4440 seconds, which is 639360KB=640MB.

Sound is actually a type of energy wave, so it also has the characteristics of frequency and amplitude, with frequency corresponding to the time axis and amplitude corresponding to the level axis. The wave is infinitely smooth and the chain can be considered to be made up of innumerable points. Since the storage space is relatively limited, in the process of digital encoding, the points of the chain must be sampled.

What is a good bitrate guide for mp3 files? Part 2

For voice recordings such as lectures or language lessons saved to waveforms, a bit rate of 32 kilobits per second (kbps) is acceptable, although 64 kbps may provide better quality, depending on the source.

At 32 kbps, the sound may sound “flat”, but that’s understandable. A 64 kbps MP3 file created from a voice recording should sound nearly identical to the original.

Desaturated acoustic music with simple arrangements should work fine at 192kbps bitrate. You can choose 256 kbps if the music will be played on a high quality device. Music that falls into this category includes folk, “boy band” songs, easy listening, and folk music. There are also works by many classic artists such as James Taylor, Linda Longstadt, Jonny Mitchell, and Simon Garfunkel.

To produce high-quality MP3 files of classical and jazz music, the optimal bitrate depends on the characteristics of the song. Smooth jazz can usually be copied at 192kbps to create a good balance between file size and diminishing returns, although 256kbps may sound better in a home entertainment center. A classical orchestra should be 256kbps for a portable player, but if you want to burn a CD at home or in your car, a 320kbps file might be a better choice.

For saturated music such as hard rock, metal, arena, pop, electronic and house music, 320 kbps will provide the best results. The higher the number of bits per second, the more complex acoustic envelope will be preserved.

If possible, it’s best to create MP3 files with variable bit rates. This allows the encoding program to determine if a particular frame of music requires the full bit rate. Otherwise, the program will reduce data retention for that frame, resulting in a smaller file without sacrificing quality. Forcing the program to “oversample” frames can produce artifacts.

While this article is intended as a general guide, he or she may be equally satisfied with a lower bitrate for a particular song or songs in general. Many factors affect our ability to judge the quality of music, not only the devices we use but also our activities while listening to it. For example, for those who listen to MP3 files while exercising or taking a walk, external noise can make it more difficult to tell the difference in quality. Conversely, audiophiles may prefer to sample at 320kbps, regardless of their equipment, type of music, or listening habits.

If you create your own MP3 files, there are other settings that affect quality. LAME is an excellent MP3 encoder and it is free and has many graphical interfaces as the interface for this popular command line program. LAME allows users to adjust many settings to generate high-quality MP3 files in seconds. You can also experiment with various bitrates in your source file to find the best subjective balance between quality and file size.

What is a good bitrate guide for mp3 files?

MP3 files are compressed audio files created from audio formats such as wave (.wav).

Wave files replicate analog recordings and digital sound files at the expense of large file size, while MP3 files sacrifice some quality for a smaller footprint. There are several factors that mitigate the quality sacrifice during the conversion process. With the correct bitrate and settings, MP3 files can provide very high quality results, making them very close to the original wave files when played on portable audio players. …
MP3 files are compressed audio files created from audio formats such as wave (.wav). Wave files replicate analog recordings and digital sound files at the expense of large file size, while MP3 files sacrifice some quality for a smaller footprint. There are several factors that mitigate the quality sacrifice during the conversion process. With the correct bitrate and settings, MP3 files can provide very high quality results, making them very close to the original wave files when played on portable audio players.

An mp3 player.
The balance between file size and quality is somewhat subjective. For audiophiles, any difference is noticeable. Others may simply not be able to tell the difference between a high quality MP3 file and a raw wave source. In many cases, the nuances of the sound environment will only become clearer when played through a high-quality stereo system.

MP3s are compressed digital music files that sacrifice quality for file size.
MP3 files are primarily targeted at portable audio players. In this field, high-quality MP3 files are played with incredible sound due to their small file size. With the limited memory of portable players, it makes sense that one would want MP3 files to be as small as possible while maintaining the highest possible quality.

For this, one of the most important factors when creating MP3 files is the bit rate. In general, the more bits per second that are preserved from the original file, the higher the quality of the MP3 and the larger the file size. Lower bit rates reduce size and quality. The idea is to use the bitrate for maximum realism without saving unnecessary data, which just creates larger files with no noticeable difference to the ear.

MP3 bitrate encoding mode

Bit rate of 1 MP3
Generally, there are three mp3 bitrates namely VBR, ABR and CBR.

1.1 RBC
CBR is short for Constant Bit Rate, which means Fixed Bit Rate in Chinese.

For a CBR MP3 song with a bitrate of 128kbps, the first 128kb of the song describes the sound of the first second, and the second 128kb describes the sound of the second second…if the song is finished, it will take 640 seconds, then the song size is 128kb × 640 = 80Mb = 10MB. The so-called 128kbps means 128kb per second.

If you are careful, you will find that the volume compressed by this encoding method will be very large, because the bit rate is fixed. Of course, the sound quality has some advantages over the other two, although this advantage may be minimal.

1.2VBR
Dynamic bit rate VBR (Variable Bitrate). That is, there is no fixed bitrate and the compression software determines on the fly which bitrate to use based on the audio data being compressed.

A simple understanding is that the bitrate will be relatively high at the time the song is rich in detail, and relatively low at other times, so sound quality and size are taken into account. For example: at the beginning of the song, a person sings alone, the sound is relatively simple, we use 64kb to describe the sound within one second; at the climax of the song, everyone sings, the sound is more complicated, we use 256kb to describe a second voice within the species.

1.3 APR
ABR (Average Bit Rate) Average Bit Rate is an interpolation parameter of VBR.

For example, when you specify 192kbps ABR to encode a wav file, Lame will use a fixed 192kbps encoding for 85% of the file, then dynamically optimize the remaining 15%: complex parts are encoded with more than 192kbps, simple parts are encoded with less than 192 kbps. Compared to CBR 192kbps, ABR 192kbps has a similar file size, but the sound quality is much better. ABR encoding is 2 to 3 times faster than VBR encoding and has better quality than CBR in the range of 128 to 256 kbps.

It can be used as a compromise between VBR and CBR. Under normal circumstances, files with this encoding method are rarely found.

The relationship between MP3 frequency, bit rate, bit rate and sound quality. Part 2

Speaking of mp3, I am afraid no one will say that they have never heard of it.

Even if you are not an mp3 user, there are ubiquitous advertisements, advertising activities in the city, discussions between friends and the Internet. Rich resources, these always give you a little impression, right? For trendy youngsters, especially friends who like music and friends who like digital devices, mp3 is probably a word that should be talked about every day, but what is mp3, how to determine mp3 sound quality and what is good or How can I listen to high quality mp3? ? ? I think the following article can help you solve many doubts.
Across current mp3 users, the generally accepted standard for production is eac recording + lame compression. Those who are experienced in such production process will figure out some tricks and use different parameter and parameter settings for different music. The compression ratio varies from the standard 128 kbps to the maximum of 320 kbps, but what is the difference and the difference in effect between these bit rates? ? How is the most suitable compression ratio, which one should be better for cbr and vbr etc. These topics are often discussed by everyone. Let me share with you some of my feelings.
The repertoire selected for this test is the first track of Bach’s “Grandenburg Concerto”, performed by the Munich Bach Orchestra, eac track recording software, cd’ex compression software, fooba2000 v0.8 playback software and listening earphones are er6 from Intech and e3c from Shure. Because the classical repertoire is very detailed and the band is large, the requirements for all aspects of sound quality are relatively high, so it can clearly reflect the difference in detail between different processing methods.
I first grabbed the track with rac, and then used the lame mp3 encoder (vision 1.92 engine 3.92) engine in the cd’ex software to process the wav file. I tried the lick parameters one by one to choose a good effect:
The first thread priority parameter selects the highest and lowest respectively. When other parameters are the same, the compression comparison shows that the degree of thread priority has no effect on the sound. The size of the generated files is the same, and the comparison sounds the same, so these parameters have no effect on the sound quality.
The second parameter is the version, which can be selected between mpegI, mpegII and mpegII.V. Similarly, the other parameters are determined and these three options are used to compress three times. After listening, although the file sizes of the three methods are all the same, but the actual listening feeling of mpegI is better. The mid-low frequency compression ratio is a bit smaller, but the high frequency distortion is a bit more. It is more suitable for listening to human voice and pop music. It is also good to use mpegI type to listen to classical music, the sound background is better, but if it is solo music with a lot of mid and high frequencies like violin, it is recommended to use mpegII.v type, which will have better results.
The third parameter is the most important, which is the bit rate. Choosing it directly affects the size and listening experience of your mp3 file. The higher the compression ratio, the higher the distortion, and the lower the compression ratio, the lower the distortion, but how do we find one for ourselves? What is the acceptable balance between the two? This requires careful exploration in the experiment. Considering that the sound quality of low bitrate files is not suitable for playing music, the minimum set is 128kbps, and four fixed bitrate files of 128, 192, 256 and 320 are used for comparison. and try.
The compression ratio of 128 kbps is still relatively rough, and the high-frequency part has obvious distortion after compression. It sounds hollow, wrinkled, scratchy, and often has a flickering sound. Misunderstanding, the compressed volume of a 3 minute 39 piece of music is 3414kb, although the volume is not large, the sound is not satisfactory, and there is a relatively large flaw.
192kbps bit rate compression effect is much better than 128. First of all, the sound is solid, at least there is no empty feeling, the high-frequency distortion is also much less, the sound is compact, the noise is small and clean, achieving relatively ideal listening. The sound effect, only because the compression is still relatively strong.