Digital Audio Quality


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Digital Audio Quality

Digital Audio Quality
Digital Audio Quality

Data rate refers to the data flow used by a video file in a unit of time, also called bit rate or bit stream rate.

Digital Audio Quality
Digital Audio Quality

The popular interpretation is the sampling rate, which is the most important part of image quality control in video encoding. Generally, the units we use are kb/s or Mb/s. Generally speaking, at the same resolution, the higher the code stream of the video file, the lower the compression ratio and the higher the image quality. The higher the code stream, the higher the sampling rate per unit time, the higher the data stream, the higher the accuracy, the closer the processed file is to the original file, the better the image quality, the clearer the image quality and the higher the decoding capability of the playback device is required.

Of course, the larger the code stream, the larger the file size. The calculation formula is file size = time X code rate/8. For example, a 720P RMVB file with a 1 Mbps stream of 90 minutes is common on the Internet and its volume is = 5400 seconds × 1 Mb/8 = 675 MB.

Generally speaking, a video file includes images and sounds, just like an RMVB video file, which contains video information and audio information. Audio and video have their own sampling methods and different bit rates, that is, the same video Audio and video file bit rate is not the same. And what we’re talking about is the bitrate of a video file, which generally refers to the sum of the bitrate of the audio and video information in the video file.

Taking the most popular and familiar RMVB video file in China as an example, VB in RMVB refers to VBR, which is short for Variable Bit Rate. The Chinese meaning is variable bit rate, which means that RMVB adopts dynamic encoding. In this way, a higher sample rate is used for complex dynamic images (singing and dancing, flying cars, wars, actions, etc.), while a lower sample rate is used for static images, and the resources are use rationally to achieve image quality and volume .Effect.

The most fundamental difference between code rate and sample rate is that the code rate is for the source file.

 

2. Sampling rate

Sample rate (also called sample rate or sample rate) defines the number of samples per second taken from a continuous signal to form a discrete signal, and is expressed in hertz (Hz). Sampling rate refers to the sampling frequency when converting an analog signal to a digital signal, i.e. how many points are sampled per unit of time. How many bits are in the data for a sample point? 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. Bit rate = sample rate x number of bits used x number of channels.

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 44.1 kHz, 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 highest digits can only be distinguished by instruments.


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Detailed Music Format Part 2

Detailed Music Format Part 2

Music Format
Music Format

Music CD

Music Format
Music Format

 

That is, CD records. A CD can play sound files of approximately 74 minutes. The Windows system comes with a CD player. Also, the software that comes with most sound cards provides CD playback functionality, and even some CD-ROM drives are offline. from computer Can be used as a stand-alone CD player when powered on.

WMA with unlimited potential

In developing its own network media service platform, Microsoft primarily promotes ASF (Audio Streaming Format), which is an open standard that supports data transmission over various networks and protocols. It supports audio, video, and a variety of other types of multimedia. And WMA is short for Windows Media Audio, which is equivalent to an ASF file that contains only audio.
The compression ratio of WMA files can be as high as 1:18 in 80kbps 44kHz mode, which is basically the same as VQF. And the compression speed is doubled compared to MP3. So it should be more competitive than VQF.

Vorbis free music format

To avoid rising royalties charged by MP3 music companies, programmers at GMGI’s iCast company developed a new free music format, Vorbis, that rivals or even exceeds MP3 in sound quality. And it will be released over the internet and can be downloaded for free without worrying about infringement issues. But MP3 has become very popular on the Internet, and Microsoft’s Windows Media technology has also started to spread, and Vorbis’s outlook is still not optimistic.

Other audio formats

AIF/AIFF: A sound file format developed by Apple, supported by the MAC platform, and supports 16-bit stereo at 44.1 kHz.
AU: SUN’s AU Compressed Sound File Format, which only supports 8-bit sound, is a commonly used sound file format on the Internet, mainly created by SUN workstations.
CDA: CD audio track file.
CMF: A MIDI-like sound file developed by CREATIVE.
DSP: Abbreviation for digital signal processing. By improving the signal processing method, sound quality will be greatly improved and songs will be more pleasing to the ear.
S3U: MP3 playback file list
RMI: MIDI Instrument Sequence

Lossy compression:

AAC – Sound quality is second only to MPC at high bit rates and looks good at both high and low bit rates. The encoding speed is too slow!
MPC: Performance is average at low bitrate, not as good as MP3 and OGG encoded by Mp3Pro, sound quality is best at high bitrate, and encoding speed is
fast.OGG: The sound quality is better at a low bitrate, and the same is true at a high bitrate. Encoding is slightly slower.
MP3 (MP3Pro): Sound quality is lower than OGG at low bit rate and other aspects are the same as MP3
WMA: High and low bit rates are average, VBR is not supported and the highest is 192Kbit/s

lossless compression:

FLAC – Worst compression ratio of the four, decent encoding speed, good platform support.
PAC: Slightly slower encoding speed, third in compression ratio, good platform support.
APE: The fastest encoding speed, the best compression rate, and the platform is generally supported.
WV: The encoding speed is very fast, the compression rate is second among the four types, and it is only supported by the Windows platform.

Detailed music format

Detailed music format

Audio File Formats
Audio File Formats

classic wave

Audio File Formats
Audio File Formats

As the most classic Windows media audio format, the WAVE file is widely used, which uses three parameters to represent sound: the number of sampled bits, the sample rate, and the number of channels.
The channels are divided into mono and stereo, and the sample rates are generally 11025 Hz (11 kHz), 22050 Hz (22 kHz), and 44100 Hz (44 kHz). The capacity occupied by the WAVE file = (sampling frequency × sampling bits × channel) × time/8 (1 byte = 8 bits).

traditional mod

MOD is a wavetable-like music format, but its structure is similar to MIDI, it uses real samples, and the volume is small. In the earlier DOS era, MOD was often used as background music for games. Modern mods can contain many audio tracks in many formats, such as S3M, NST, 669, MTM, XM, IT, XT, and RT.

midi music computer

MIDI is short for Musical Instrument Data Interface. Records the sound played by the instrument digitally (each note is recorded as a number), and then synthesizes these records via FM or wavetable during playback: FM synthesis is the sound of the instrument is simulated by mixing the multi-frequency sounds; wavetable synthesis consists of storing the sound samples of the instrument in the wavetable of the sound card and extracting the sound from the wavetable as you play.

Boss Boss MP3

It can be said that MP3 is famous, it uses MPEG Audio Layer 3 technology to compress the sound with a compression ratio of 1:10 or even 1:12, with a sampling rate of 44kHz and a bit rate of 112kbit/s. .
MP3 music is music stored in digital form. If you want to play it, you must have a corresponding digital playback and decoding system. Generally, MP3 digital music is decoded by special software and then restored to a waveform sound signal for playback output. This type of software is called For MP3 players, such as Winamp, etc.

Overlord RA series online

RA, RAM, and RM are Real’s mature network audio formats, using “streaming audio” technology, making them well suited for network streaming. Information such as copyright, singer, producer, mail and song title can be added during production.
RA can be called the supreme lord of multimedia communication on the Internet. It is suitable for streaming on the Internet and is currently the best format for listening to online music online.

VQF with high compression ratio

VQF or TwinVQ is an audio compression technology developed by Nippon Telegraph and Telephone and Yamaha Corporation.
The audio compression rate of VQF is almost twice that of standard MPEG audio and can reach approximately 1:18 or even higher. And popular compression formats like MP3 and RA are usually only around 1:12. But it still won’t affect the sound quality, when VQF compress music at 44kHz-80kbit/s audio sampling rate, its sound quality will be better than 44kHz-128kbit/s MP3, when compress at 44kHz-96kbit/s , the music is close to 44kHz-256kbit/s MP3.

MD minidisc

MD (ie MiniDisc) is a comprehensive portable music format released by SONY in 1992. The compression algorithm it uses is ATRAC technology (the compression ratio is 1:5). MD is divided into Recordable MD (Recordable, with two heads of magnetic head and laser head) and Single Play MD (Prerecorded, only laser head).
The powerful editing function is the strong point of MD. You can quickly select tracks, move tracks, merge, split, delete and edit track titles. It is more personalized than CD and you can have your own MD album at any time. MD products include MD Walkman, MD bedside audio, MD car audio, MD recording deck, MD camera gun and MD driver, etc.

What is the AC3 format?

The AC3 format, derived from Audio Coding 3, is an extension to surround sound files. It was created by Dolby Laboratories in 1987 for use on DVDs, Blu-ray players, HDTV programming and home entertainment systems. The AC3 format contains up to 6 discrete audio channels. The 5 most frequently used channels are dedicated to the normal range of speakers (from 20 to 20,000 Hz) and 1 low frequency channel (from 20 to 120 Hz) subwoofer power. In particular, the front, front right, middle, rear left, rear right and 1 ultraviolet track are called 5.1 channel, which is the common surround sound most commonly used in commercial rooms and home theaters.

ac3

AC3 format

File type name: AC3

Category: Audio Files

Popularity: Popular with home theater enthusiasts

Developer: Dolby Laboratories

AC3 format information

ac3

The AC3 format can operate below the audio frequency from 20 to 20,000 Hz, which corresponds to the frequency heard for the human ear. This means that AC3 can produce a unique, detailed and varied sound effect for humans.
Most HDTV programming today uses AC3 as the standard audio format. The combination with the television broadcast’s HD signal makes everything realistic, especially the sound.
Previously, it was not adopted as the standard format for the AC3 company, but Dolby Pro Logic 4.0. Dolby Pro Logic 4.0 can be found on many VHS tapes released in the 1980s.
The bit rate of AC3 audio on DVD reaches up to 640 Kbps and the sampling rate up to 48 kHz.
Benefits: AC3 is accepted as an industry standard for DVD and DTV media. Almost all the audio tracks for the DVD movies take place in AAC format. AC3 is very compressed and is a small file, but the sound effect is true to the sound of the original DVD. AC3 can be converted to video files such as AVI and MPEG to obtain 5.1 Dolby Digital Surround audio or for playback in media players.

Cons: AC3 is widely used on DVDs and very rarely has it appeared in other aspects. Although AC3 supports 7.1 channels when it comes to audio CDs, it only supports a maximum of 5.1 channels and a limit of 448 kbps.

AC3 VS. AAC

AAC or Advanced Audio Coding is very different from AC3. AAC is a type of lossless digital audio format that is promoted as a successor of MP3 because it generally achieves better audio quality within the same file size. The AAC format is a more advanced lossless format than the AC3 format. It will create better sound quality than AC3 with the same bitrate, especially at low bitrate. Below you will find a more in-depth comparison between AC3 and AAC.

History:

AC3 was developed by Dolby Laboratories and was originally called Dolby Digital Stereo until 1994. This format has been taken as an international standard by the Moving Picture Experts Group since 1997.
AAC was developed in collaboration with AT&T Bell Laboratories, Dolby Laboratories, Sony Cooperation, Nokia and Fraunhofer IIS.

AC3 digital audio

AC3 digital audio

AC3 Filter is a high quality audio filter decoder and processor. Allows multimedia players to play movies with AC3 and DTS audio tracks. It can also process any audio track, much of the processing options allow you to adjust the sound in almost any way. It is possible to upmix any audio source to 6 channels. Filter can do multi-channel and digital (SPDIF) output. You can encode any audio source to AC3 on the fly and send it via SPDIF to the receiver.

AC3 Files Technical Details

AC3 File is a DirectShow source filter. It is necessary to read all kinds of AC3 or DTS files. There are many types of AC3 and DTS files. Some files are ripped from multiple audio CD channels, some files are ripped from DVD in a PES container, some files may have different byte order. All these files are AC3 File compatible. AC3Filter Tools is a set of command line tools for manipulating audio files. Includes AC3 encoder and decoder, SPDIF format converter, parametric filter, graphic equalizer, and more.

AC3 operation

AC3 is one of the so-called perceptual compression formats. What it basically does is remove all parts of the original, analog-encoded sound that cannot be perceived by the human ear. In this way, it is achieved that the same information is smaller and therefore occupies much less physical space.

AC3 Development

The popular audio encoder developed by Dolby Labs and which was released in theaters almost 20 years ago with the movie Batman Returns. This technology, also known as Dolby Digital, was designed precisely to enrich the audio systems of movie theaters, and little by little it has been incorporated into the home, thus providing high-value experiences for Home Cinema systems.

The AC3Filter codec allows you to decode Dolby Digital AC3 sound, which is a type of 5.1 channel surround sound (although there is also a 2.0 channel stereo version). This very good quality sound is normally included in most DVD movies. Audio tracks in AC3 format can be found not only on DVDs, but also in movies compressed into AVI, MKV or MP4 files.
AC3Filter not only supports AC3 sound, but also supports DTS sound. It is a matter of activating the corresponding option in the codec configuration.