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Digital Audio Quality

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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.

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

Music CD

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

classic wave

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.

Compressed audio formats

Compressed audio formats

Compressed Audio File Formats

Understanding compressed audio formats
The digital age dictates its own laws, according to which, in particular, audio and video information is more convenient to store and transmit in compressed form. Let’s briefly discuss the principle of sound compression.

Compressed Audio file formats

As you know, the music we listen to consists of a set of signals, each of which has its own characteristics, including loudness. The human auditory system is designed so that we do not distinguish or misdirect a weak (low) signal from the background of a strong (strong) signal. This principle forms the basis of modern means of compression (compression) of audio data.

If we imagine that a signal of a certain length is divided into many parts, and each part is processed in such a way that a weaker signal, which is difficult to distinguish from a strong one, falls under the knife and a stronger signal remains, then this will be a rough model of audio signal compression. … Consequently, the level of data compression will depend on how many parts (samples) the original file will be divided into and how many weak signals from each individual sample will be removed (what the bit rate will be: the number of bits in a sample of a specified duration).

The first versions of codecs for data compression acted quite crudely: they just cut off a weak signal and did not take into account the type of music, therefore, rather energetic music, without special nuances, in a compressed form does not it sounded worse than the original, whereas more complex classical and acoustic music simply lost all color and depth.

As a result of this, a transition to a more intelligent compression algorithm, with a variable bit rate, was made. Depending on the musical texture, that is, the ratio of weak and strong signals, the codec changes the amount of weak signals cut, so that we hear a more believable sound.

Obviously, with a higher sample rate (sampling) of 44.1-48.0 KHz and a higher bit rate (160-192 Kbps), we will get a sound more consistent with the original than with a sample rate 22 KHz and 64 Kbps bit rate. However, the size of the final compressed file is directly proportional to the selected sample rate and bit rate, and this is what people who distribute music in the form of compressed (compressed).

It should also be remembered that most algorithms cut the upper part of the audible range as well, starting at around 15 kHz.

There are currently several original compression algorithms, most of which are compatible with Linux.

Ogg Vorbis
Ogg Vorbis is a completely open audio format that allows you to store and transmit audio information with high sound quality (44.1-48.0 kHz sample rate, 16+ bits, polyphony (multi-channel audio)) and bit rates ranging from 16 to 512 kbps per channel. The number of channels processed can be as high as 255. This allows Vorbis to be on par with MPEG-4 (AAC and TwinVQ), WMA and PAC audio, and clearly superior to MPEG-1 Layer 3 (MP3) audio. .

Ogg Vorbis is also a streaming format, allowing it to be used, for example, for Internet broadcasts, especially since this format is compatible with Icecast. The characteristics of the codec algorithm allow you to get the final file smaller than MP3 files of similar quality.

For the reproduction the console program ogg123 is used, to encode – oggenc; both have graphic housings. More details on both are in the following sections.

MP3
MP3 or MPEG-1 audio layer 3 is by far the most popular format for storing and transmitting compressed data. This format was developed by the Frauenhofer Institut, Germany. However, despite the ubiquity of the format, it should not be forgotten that the patent for MP3 encoding and decoding algorithms belongs to a single company, so the end user at any time may find themselves in a very disadvantageous environment, such as It has already happened with the developers of free MP3 data compression tools …. You can get details about the license conditions on the developers website.

WMA
The WMA format is a proprietary product of Microsoft. It failed to occupy a market segment comparable to MP3, but it has some popularity despite serious security concerns identified. At the moment, only the universal MPlayer player can play WMA files. There are no free data compression tools for this algorithm and its appearance is unlikely.

Main parameters to guarantee the audio quality of your digital product

Discover the main parameters to guarantee the audio quality of your digital product

Do you know the differences that exist between the formats? Do you understand how the compression rate works?

Audio settings for recorders or sound interfaces can be very confusing. But, if you are going to work with videos or podcasts, it will be useful to know how to interpret the parameters when recording and exporting files, either in Audacity (free), Reaper, Adobe Audition or in video editors.

Here we are going to talk about the differences between sampling rates (sample rate), resolution (bit depth), file compression rates and format variations. Thus, you will be more sure of the options you have regarding audio quality and you can guarantee good results.

In short, you will understand why we recommend recording in uncompressed format (WAV, for example) in 24 bits and 48 kHz. In addition, you will also know why, in most cases, we do not need more than a 192 kbps MP3 to export excellent quality audio.

We will also talk about the possibility of compressing more podcast files, which can be generated in MP3 64 kbps, mono, to facilitate online consumption.

Formats, extensions and codecs: What do they mean?
When it comes to audio files, we can talk about formats, extensions and codecs. In summary, we can say that the format refers to the type of file, identified by its extension (* .mp3, * .wav, * .ogg, * .wma etc), which often tells us how it has been encoded or which It is your codec.

For example, a file in the MP3 format has * .mp3 extension and MPEG-1 Audio Layer III codec.

Examples of audio file extensions

Normally those endings are mixed. But what is important to know is that, as in videos, files with the same type of extension do not always have the same codec and vice versa.

That information is valid so that you do not feel lost in case you do not understand the reason why a software, which normally plays your * .m4a files, does not play another one with the same extension, for example.

Such a situation could indicate that the codecs used are different. In that case, the solution would be to use other software to read the file or to convert it (new encoding). This can be done even in video editors.

The variations of formats and codecs depend on the options of the companies that develop the software that executes the files. In these cases, there are many things at stake, such as technical specifications and relations with patents.

On the other hand, files are usually divided into two types: without compression or compressed.

Files without compression

Audio recording equipment usually offers us options to record files without losing any information. These files, not compressed, can be generated in various formats and extensions, such as WAV, AIFF, FLAC and ALAC. For those who are familiar with photography, they are equivalent to RAW or DNG.

As they are usually very heavy, using lossless formats in the final product is only recommended in some cases, such as:

when the final product can be processed by the consumer (files intended for sound banks, for example);
when there will be recording on physical media (CD, DVD and Blue-Ray);
or for the audiophile market (for a matter of perceived value and high quality assurance).
But, even if you don’t want to end the process with a WAV (one of the most common), lossless formats can be very useful in the editing stage. Because they contain a lot of information, they support more extreme alterations without harming the audio quality.

With plugins, conversions and processing, they can be handled more freely, guaranteeing excellent quality, even if a compressed file is subsequently generated.

Compressed files

Most of the equipment available in the market (cameras, cell phones and even audio recorders) usually delivers already compressed files. This type of file is more practical, easy to process, requires less storage space and has very small sizes (in bytes).

Some examples of these formats are: 3GP, AAC, M4A, OGG, WMA and MP3, which is, without a doubt, the best known. The files are like JPEG or GIF in the field of images.

Through a complex algorithm, these files are generated seeking to keep only relevant information for our ears. Depending on the compression mode, we can generate an MP3 from a WAV and have a file 10 times smaller, without noticeable changes in audio quality.