Is it lossless after converting mp3 format to wav format?
WAV vs MP3
Lossless, that is, lossless compression, is a relative concept, relative to the concept of the compressed source file. Usually this source file is a wav file.
The size of the wav file is based on the sample rate, the quantization precision, and the number of channels. The mp3 file is obtained by cutting off some of the high frequencies that are not easily audible to the human ear in the wav file. The mp3 file is converted to a wav file, even if the trimmed part is resampled, unfortunately the information here disappears and the resampling is empty. wav is equivalent to a 1000ml beverage bottle, a 1000ml bottle. The mp3 is equivalent to condensing 500ml of it, leaving only 500ml in the bottle. Converting mp3 to wav is equivalent to putting a 500ml concentrated drink back into a 1000ml bottle.
2. The higher the bitrate, the better the sound quality?
From a general point of view, the higher the bitrate, the better the sound quality. But the sound quality is the same for the fake lossless conversion from mp3 format to wav format. There is another piece of information that is also related to sound quality, which is the compression rate; the higher the compression rate, the worse the sound quality.
Both MP3 files and WAV files are digital audio formats.
WAV vs. MP3
Due to differences in encoding and compression ratio, the two have different file sizes and sound quality. WAV is the closest music format to lossless, and MP3 files save space by encoding the audio and removing certain parts.
Difference 1: different compression ratios
WAV is a type of lossless music. The data format itself is PCM or compressed, which is a kind of lossless music format. So WAV files are larger than MP3 files;
MP3 is an audio compression technology, its full name is MPEG Audio Layer 3 (MPEG Audio Layer 3), abbreviated as MP3. It is designed to drastically reduce the amount of audio data. MP3 converts the time-domain waveform signal to a frequency-domain signal by using the human ear’s insensitivity to high-frequency sound signals and dividing it into multiple frequency bands. increased for high frequencies (even ignoring the signal) Use a small compression ratio for low frequency signals to ensure that the signal is not distorted. In this way, it is equivalent to discarding the high-frequency sound that is basically inaudible to the human ear, and only retaining the low-frequency part that can be heard, thus compressing the sound with a compression ratio of 1:10 or even 1:12. This compression method is the MPEG Audio Player3 mentioned above.
Difference 2: the sound quality is different
Due to the difference in the degree of compression, the sound quality of the two is different. WAV is the closest lossless music format. WAV file with standard format is the same as CD format, and it also has 44.1K sample rate and 16-bit quantized numbers. Therefore, the sound file quality is almost the same as CD;
MP3 files save space by encoding the audio, removing some parts, but there is no difference between normal listening and normal equipment and normal volume.
Sound has three elements: pitch, volume, and timbre:
WAV vs MP3
Pitch is determined by the frequency of the sound wave, the higher the frequency, the higher the pitch.
The volume is determined by the amplitude of the sound wave, the larger the amplitude, the louder the sound.
The timbre is determined by the “shape” of the waveform (sounds like square, triangle, and sawtooth are called impulse waves and sound individual).
An audio file is a file obtained by converting an analog signal to a digital signal. In general, there are 5 important parameters: encoding method, number of channels, sampling rate, bit depth, and bit rate.
Encoding: how this format organizes binary data and how it is compressed.
Number of channels: mono, dual or 5.1 channels, etc.
Sampling rate: The number of samples per second.
Bit Depth: The number of binary bits used to store the y value of the sample point.
Bitrate – The desired number of bits per second for the file.
We know that there is no compression in the WAV format, so its encoding method is to directly write all the sampled points to the file in order.
WAV file size (B) = number of channels * sample rate (Hz) * bit depth (bit) / 8 + the file header size (B, it’s 44B)
Implementation
When you open an mp3 or wav file with a text editor, you see numbers like this:
4944 3303 0000 0000 3d48 5459 4552 0000
0006 0000 0032 3031 3800 5444 4154 0000
0006 0000 0032 3230 3300 5449 4d45 0000
0006 0000 0031 3430 3600 5052 4956 0000
168e 0000 584d 5000 3c3f 7870 6163 6b65
7420 6265 6769 6e3d 22ef bbbf 2220 6964
3d22 5735 4D30 4D70 4365 6869 487A 7265
537A 4E54 637A 6B63 3964 223F 3E0A 3A78
6D70 6D65 7461 2078 6D6C 6E78 3D22
6F62 653A 6574 612F
5249 4646 2e3d 0e05 5741 5645 666d 7420
1200 0000 0300 0200 44ac 0000 2062 0500
0800 2000 0000 6461 7461 a026 0e05 8089
00bc 00e8 f0bb c09e 8dbc 00c2 87bc 80f1
d3bc 8063 ccbc c030 fcbc 8012 f4bc 20bb
13bd e051 0fbd c0b0 2dbd 6079 28bd 4012
46bd 6032 40bd c0e3 5dbd 6040 57bd c015
7cbd e035 74bd b058 8dbd 50e2 88bd f0a7 9dbd e0dd 98bd 70d3 acbd e0a9 a7bd
d043 b8bd b0da b2bd
00e3 c4bd 605c bfbd
This one above is the mp3/wav format of the same song. What is the difference between them?
WAV
structure
file header
The WAV format follows the RIFF Resource Interchange File Format, so the WAV format is actually a three-layer relationship, which is simplified here. Its file header format is as follows:
Address Carving type content
00H-03H 4 character * 4 RIFF resource file exchange flag
04H-07H 4 unsigned int The number of bytes from the next address to the end of the file.
08H-0BH 4 character * 4 WAV file WAVE logo
0CH-0FH 4 character * 4 fmt wave file flag, the last digit is 0x20 space
10H-13H 4 unsigned int The size of the subchunk file header. For the WAV subfragment, the value is 0x10.
14H-15H 2 short unsigned Format type, when the value is 1, it means the data is linear PCM encoding
16H-17H 2 short unsigned number of channels
18H-1BH 4 int unsigned Sampling rate
1CH-1FH 4 int unsigned Wave file bytes per second = sample rate Bit depth PCM / 8 channels
20H-21H 2 short unsigned DATA data block unit length = number of channels * PCM bit depth / 8
22H-23H 2 short unsigned Bit depth PCM
24H-27H 4 character * 4 data stamp data
28H-2BH 4 unsigned int Total length of data part (bytes)
struct WAVHeader
{ char RIFF[ 4 ]; ///Resource file exchange flag RIFF unsigned LEN; ///Number of bytes from the next address to the end of the file char WAV[ 4 ]; ///WAV file flag WAVE char FMT [ 4 ]; ///Wave fmt file pointer, last digit is 0x20 space unsigned SubchunkSize; ///The size of the sub-chunk file header, for WAV this sub-chunk, the value is 0x10 DATATYPE short unsigned; / //Format type, when the value is 1, it means the data is unsigned linear PCM encoding short CH ; ///Number of unsigned channels F; ///Unsigned sample rate BYTERATE; ///Number of bytes per second of wave file = sample rate*PCM bit depth/8*Number of unsigned channels
short DATAUNITLEN; ///DATA block unit length=channel number*PCM bit depth/unsigned 8 short BITDEPTH; ///PCM bit depth character DATA[ 4 ]; ///Unsigned data mark data DATALEN ; ///Total data section length (bytes) };
44100Hz represents the sample rate of the signal. The so-called sampling consists of obtaining the value y of the sound wave at the current moment every unit of time. Sampling is the process of discretizing continuous data (converting an analog signal to a digital signal).
image source
The sampling method mentioned above is called PCM (Pulse Code Modulation). According to the Nyquist-Shannon sampling law, the sampling rate must be at least twice the highest target frequency. The hearing range of the human ear is about 20Hz-20,000Hz (if you’re curious how loud you can hear, you can click here to test your ears), although recording software often has a 48,000 option Hz, but we can safely conclude: 44100Hz can meet almost all our needs, higher is just a waste of your memory and CPU. More than 48,000 samples are meaningless to the human ear, which is similar to 24 frames per second on a movie. 44100Hz happens to be the standard sample rate for almost all music released. In fact, for vocals and many instruments, high-frequency sounds are noise, so high sample rates can sometimes worsen sound quality (which is why we need to adjust the equalizer).
320kbps represents your bit rate/bit rate, which is short for kilobits per second, which represents the size of the data used to describe sound. In CD (uncompressed audio file), the bit rate is 1411.2 kbps, and the mp3 sound quality to achieve CD quality should be higher than 128 kbps / 44100 Hz (128 kbps can be said to be the most common bit rate). Generally, a higher number means better quality. The quality depends on many factors (such as the encoding algorithm). Many times we don’t need too high bitrate: our device can play mp3 and CD without difference (sound/sound card is normal).
A wav is 44100 Hz 16-bit stereo or 22050 Hz 8-bit mono, what does this mean? stereo/mono refers to dual/mono. For monophonic sound files, the sample data is an eight-bit short integer (short int 00H-FFH); for two-channel stereo sound files, each sample data is a 16-bit integer (int) and the upper eight bits (left channel) and lower eight bits (right channel) represent the two channels, respectively.
Sound is a mechanical wave, produced by the vibration of an object, and requires a medium to propagate. So, in essence, a sound is a waveform on an axis over time.
Comparing the difference between AAC and OGG, two lossless audio formats
AAC vs Ogg
what sound quality will be better?
AAC vs Ogg
Anyone who likes to listen to songs should know that the sound quality of different audio formats is also different. Do you know the difference between the two audio formats of AAC and OGG? Let’s find out together.
1. AAC Features
1. AAC is characterized by high sampling rate and multi-channel. Compared to MP3 and CD, the sample rate is 44khz. Compared to AAC, the sampling frequency is more than 90khz, and AAC is also divided into multiple files, so the sound quality of AAC will be higher.
2. If you are only looking for sound quality, the higher the sound quality, the better. Everyone knows that the highest AAC mobile playback device is almost 320. If you want more, it is better to choose lossless. AAC can almost replace MP3.
Second, the features of OGG
1. OGG is characterized by open source and free and open restrictions, very similar to MP3, but its volume is smaller than MP3, and it also supports multi-channel. And if you want to change its sound quality, it won’t affect the original encoding and the player won’t change.
2. It can accommodate many types of open source code, including audio, video and text metadata processing. If the output is selected, q0, q2, q4 and q10 are the best sound quality materials. Many people who have sound quality requirements want to download lossless music.
If you want to download lossless music, you need to be a member. If you want to go to a higher level, you can convert it to listen to it, which is also a good option. For example, use the fast audio converter to convert the format, whether it is FLAC format, WAN format or MP3. Whether you want to rip or cut audio, merging is also supported.
In fact, listening to music is not that important in terms of sound quality or other aspects. It mainly depends on personal feelings. Of course, it’s understandable that every search is different, so if you have better suggestions or want to add more. you can leave a message to discuss.
What does the level 1.10 ogg mean, sorry, I don’t quite understand this. If it is q0, q2, q4 in the output options, the q10 has the best sound quality.
2.192k, which is the code rate, also known as the bit rate, and the sampling rate mentioned above are two concepts. The higher the bitrate, the better the sound quality, but the larger the file size.
3. Aac generally supports up to 320 for mobile playback devices. It seems that the use of Qianqianjing on the PC can reach 440 kbps. If you’re just looking for sound quality, the higher the bitrate, the better. However, I personally think that there is no point in going above 320. It is better to listen directly to lossless ones.
4.aac, aac format is designed to replace mp3, but patent rate limitation makes this format still not popular. ogg is less suitable for individuals I often see this format in some game installation folders.
When it comes to downloading music, the MP3 audio file format used to reign supreme.
In fact, this format is so synonymous with mobile music solutions that “MP3 players” are now the common format for audio playback devices.
However, for various reasons, it is less prominent these days. However, it continues. Understanding MP3 files can also help us understand other formats more easily, so we’ll start there.
An MP3 file is a lossy audio file, which means that it discards data that our ears cannot hear. Almost everyone has a hearing range between 2oHz and 20kHz. The upper limit actually decreases with age, but generally speaking, it’s a lie within the range of every noise you hear. Since we know that other frequencies are redundant, MP3 discards all frequencies outside this range.
To save more space, MP3 files use some more tricks. Audio engineers use noise modeling algorithms based on the psychoacoustic effects of the human ear and brain to remove parts of music that we shouldn’t be hearing. For example, the brain cannot distinguish between two frequencies that are next to each other. Also, adult ears have difficulty recognizing the direction of high-frequency sounds. It also starts to lose sensitivity above 16kHz. Also, loud sounds can mask quieter sounds. All of these can be removed, with little noticeable difference in final audience.
Basically, MP3 files remove frequencies that we can’t hear and frequencies that we can hear individually, but not because of how they’re combined in a particular song.
MP3 divides the track into 576 sample frames and uses the Fast Fourier Transform (FFT) to obtain frequency data from these frames. The frequency data is then analyzed to see if there is any opportunity to apply compression rules based on human hearing as described above. If so, these parts are rounded down (quantized) to reduce the bitrate, which helps save space. Data on how to restore each frame to its full sonic representation is stored in a 32-bit header.
The bitrate determines the maximum file size allowed per frame. The more aggressive the compression, the more likely the algorithm will remove things that are audible. Also, this type of filtering and cutting is not perfect, and quantization can leave artifacts that some people may hear. This lossy psychoacoustic compression is followed by lossless Huffman encoding compression similar to .zip files to save even more space.
Digital audio files are a way of recording how content is saved on a computer, media player, smartphone, or other device.
audio file format
At its most basic level, digital audio is a series of numbers that a device can use to reproduce sound waves. There are several ways to accomplish this and then compress (or not) the resulting data. We know that by sampling the sound waves during analog-to-digital conversion to at least 16 bits at 44.1 kHz, we can perfectly reproduce the captured signal later on. This is thanks to some math called the Nyquist-Shannon sampling theorem. We can achieve higher bit rates and frequency ranges, but whether anyone can hear the difference, even with the best headphones, is debatable at best.
If we just kept that data as it is (called Pulse Code Modulation or PCM), the file would take up a lot of space. This is why lossy and lossless forms of audio compression were developed. Lossy audio throws out audio frequencies our ears can’t hear, while lossless audio keeps them. Lossy audio formats can also use other tricks to further compress the audio, which we’ll cover later.
Since most people access their music through streaming services these days, compressed lossy file formats are the primary way to distribute content. It doesn’t matter what you listen to, but some people demand the highest quality. As a result, more and more high quality and even lossless streaming options are now available. However, as the image below shows, there is no avoiding the fact that lossy formats take up less space and consume less mobile data.
Which is better between FLAC format and ogg format?
Positionable: FLAC supports fast sampling and precise positioning.
This is not only beneficial for playback, but also makes FLAC files easier to edit.
Resilient metadata: New types of metadata blocks can be defined and implemented without affecting the use of old decoders and data streams. Existing metadata types include tags, reference tables, and ranking tables. Registered applications can define their own dedicated metadata types (Note: this is similar to the MIDI standard).
Great for archiving applications: FLAC is an open and lossless encoding format, you can convert it to any other format you need. In addition to CRC and MD5 marking of each data frame to ensure data integrity, flac also provides a check (verify) option, when using this option for encoding, the encrypted data will be immediately decoded and compared with the data. original input in the same encoding time.
Easy to backup CDs: FLAC has a “reference table” metadata block that contains the list of CD contents and index points for all tracks. You can save a CD into a single file and import the CD reference table, so that a FLAC file can record all the information of the entire CD completely. When your original CD is damaged, you can use this file to restore an exact copy of the original CD.
Damage resistance: Due to the structure of the FLAC frame, once the data stream is damaged, the loss will be limited to the damaged data frame. Usually only a short segment is lost. When many other lossless audio compression formats suffer corruption, one corruption will cause all subsequent data to be lost.
OGGVorbis is a lossy compression encoding for audio.
FLAC is short for Free Lossless Audio Codec, which can be interpreted as lossless audio compression coding in Chinese. FLAC is a well-known free audio compression codec, which is characterized by lossless compression.
Lossless Compression: The encoded (PCM) audio data is lossless and the decoded output audio is the same in every byte of the encoder input. Each data frame has a 16-bit CRC check code of the current frame, which is used to check for data transmission errors. For the entire piece of audio data, an MD5 tag for the original uncompressed audio data is also stored in the file header, which is used to verify the data during decoding and proofing.
Fast: FLAC pays more attention to decoding speed. Decoding requires only integer operations and has low computational speed requirements relative to most encoding methods. Real-time decoding can be easily achieved on very common hardware.
Hardware support: Since FLAC provides a free decoding example and the decoding complexity is low, FLAC is currently the only lossless compression codec that has hardware support.
Streamable: Each FLAC data frame contains all the information needed for decoding. The current frame is decoded without reference to the data frame that precedes or follows it. FLAC uses sync and CRC codes (similar to encoding formats like MPEG) so that the decoder can have minimal time delays when it locates jumps in the data stream.
The Free Lossless Audio Codec is a compression standard originally developed by the non-profit foundation Xiph.org that supports digital audio files that are acoustically identical to the source material. FLAC-encoded files, usually with the .flac extension, are noted for their completely open source structure, as well as their small file size and fast decoding times.
FLAC files are very popular in the world of lossless audio. In digital audio, a lossless codec is a codec that does not lose important signal information about the original analog music during file compression. Many popular codecs use lossy compression algorithms, for example the MP3 and Windows Media Audio standards, which lose some audio fidelity during rendering.
CLOSE
rip music cd
In fact, many users who want to back up their original audio CDs (CD ripping) choose to use FLAC to preserve their sound rather than a lossy format. Doing this ensures that if the original source is damaged or lost, a perfect copy can be played using the previously encoded FLAC file.
Of all the lossless audio formats available, FLAC is probably the most popular today. In fact, some HD music services now offer tracks in this format for download.
Ripping an audio CD to FLAC generally produces a file with a compression ratio between 30% and 50%. Due to the lossless nature of the format, some people prefer to store their digital music library as FLAC files on external storage media and convert them to a lossy format (MP3, AAC, WMA, etc.) when needed, such as syncing with an MP3 player or other types of portable devices.
FLAC Property
All major operating systems support the FLAC standard, including Windows 10, macOS High Sierra and later, most Linux distributions, Android 3.1 and later, and iOS 11 and later.
FLAC files support metadata tags, album art, and quick search content. Because it is a non-proprietary format and its core technology is licensed royalty-free, FLAC is especially popular with open source developers. In particular, FLAC’s fast streaming and decoding capabilities make it suitable for online playback compared to other formats.
