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Compression audio encoding Part 3

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Compression audio encoding Part 3

I often hear what is called Hi-Res Audio. The sampling frequency is said to be 96 kHz or 192 kHz, which is over 48 kHz, the number of quantization bits is 24 bits, and the limit (high range) of human hearing is about 20 kHz, but it expresses frequencies higher than that. It will be. It is the same bit rate as the image from a long time ago. .. ..
By the way, it seems that dogs can hear up to 60 kHz and cats up to about 64 kHz.

Hi-res audio example
Sampling frequency Number of quantization bits Number of channels bit rate Frequency that can be expressed
192 kHz twenty-four 2 9.216 kbps 96 kHz
192 kHz 16 2 6,144 kbps 96 kHz
96 kHz twenty-four 2 4.608 kbps 48 kHz
96 kHz 16 2 3,072 kbps 48 kHz
48 kHz twenty-four 2 2,304 kbps 24 kHz
Considering the limit of human hearing (about 20 kHz), according to the sampling theorem, 48 kHz or 44.1 kHz is a sufficient frequency, but what about all of them? .. ..
In my case, I cannot distinguish the high resolution range, but it should be able to reproduce the discarded frequency at 48 kHz to 96 kHz, and when the number of quantization bits is in the 24-bit range, the sound pressure (dB) is a bit. Feels like I’m going up (?) (It’s just a story from my ears).
I’d like to make a comparison if I get the chance, but I don’t think I can tell by ear without a proper regenerator (like an expensive analog amp).

Is it time for cats and dogs to get verified in the acoustic industry? .. ..

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Compression audio encoding Part 2

16-bit monaural PCM bit rate (for audio) (example)

Sampling frequency Number of quantization bits Number of channels bit rate Comments
32 kHz 16 1 512 kbps Super Wide Band
24 kHz 16 1 384 kbps
16 kHz 16 1 256 kbps Broadband
8 kHz 16 1 128 kbps Narrowband

Sampling rate
If you check the web, there are explanations like the sampling required to convert analog waveforms to digital conversion. For example, it shows how many samples of an audio signal input from a microphone are taken per second and digitized. The larger the sample, the greater the range that can be recorded. When an analog waveform is digitized, the frequency that can be expressed is half the sampling frequency (sampling theorem). For example, with a sampling frequency of 48 kHz, it can be expressed up to 24 kHz. At 8 kHz (narrow band) and 16 kHz (wide band), which are often used for audio, you can only hear up to 4 kHz and 8 kHz, respectively. The higher the sample rate, the higher the bit rate.

Sampling theorem
It is a very simple explanation, but it can express up to half the sample rate. When sampling a signal, if the interval is small, it can be restored close to the original signal, but if it is too thick, it cannot be restored (I would like to write a little more detail when I talk about signal processing or other time ).

44.1 kHz
Why is there a poorly separated rate of 44.1? .. ..
Isn’t the technician deliberately wearing an annoying watch to prevent music CDs from being easily copied? I heard something like that. When I searched, it seems this happened (?) Due to the convenience of an old PCM recorder. In this age, it is difficult to know what 44.1 kHz is in development. The 44.1 kHz ↔ 48 kHz sampling conversion is a headache. For example, USB audio (USB audio device class) exchanges data at 1 ms intervals. In the case of 48kHz, the data is 48 samples, but when considering 44.1kHz, it will be 44 samples (x9) and 45 samples (x1) in 10ms. If a sample of 45 samples is misled (tentatively), it will be 44.0kHz. I think it’s more like that with voice and music, and the human ear is mostly misleading (just my personal opinion).
However, the objective evaluation method will soon come to an end. For example, you can clearly see that you were fooled by a sine wave (sine wave) (maybe you are unexpectedly on the market).

Number of quantization bits
Sampling had to take a value in the direction of time (discretization), but quantization had to take a value in the direction of amplitude. The range that is possible to display the volume of the sound, which is heard often, “dynamic range 96 dB” means that the number of quantization bits is 16 bits and the music signal is played in the range of 0 to 65535 I can do it. The number of quantization bits is also called the bit depth or bit depth.

Bitrate
In communication, it indicates how many bits of data are transferred per hour and is generally expressed in bps (bit / s) of how many bits are transferred (processed) per second. If it is small, the size when saving as a file is small and there is space on the transmission line for communication. For example, when an audio (1 channel) is compressed to 1/3, the 3 channel audio can be sent at the same bit rate. Excuse the old story, but considering from the age of analog communication (analog mobile phone), digitization + compression will be able to support multiple calls with the same radio wave.

compressing using audio encoding

When compressing using audio encoding (AAC, MP3, etc.), the compression rate is determined by the bit rate at the time of encoding.

Specifically, if you set a low bitrate, the compression rate will be high and the file size when saved will be small, but what is the bitrate for the original sound source (PCM) without compression in the first place?

If you save it as PCM, the sound quality of the original sound will be obtained, but it can be a little inconvenient to save it without worrying about the file size. Also, depending on the application, I think the original sound size has enough memory capacity and the communication speed is correct. Therefore, I would like to write about the sample rate and bit rate that are often heard in digital audio.

The bit rate of digital audio is determined by the sample rate, the number of bits assigned to a sample (number of quantization bits), and the number of channels (stereo, monaural, etc.).

PCM bit rate (uncompressed) = sample rate x number of quantization bits x number of channels
As I wrote a bit last time, file containers like wav and mp4 format have this information as the header, so the application can see the header and play it. The compression rate of the encoding is determined by the bit rate specified at the time of encoding for this PCM (uncompressed) bit rate.
For example, as many of you know about music CDs, with 44.1 kHz stereo, this is the next bit rate.

Music CD bit rate: 44100Hz x 16bit x 2ch (stereo) = 1411.2kbps
When encoding this with MP3, AAC, etc., you will naturally specify a bitrate less than 1,411.2 kbps. For example, when encoding at 256 kbps, the compression rate is approximately 18% and the file size is 1/5 or less, assuming the original sound is 100%.

Encode 256 kbps music CDs: 256 kbps / 1,411.2 kbps = approximately 18%
Generally, the sample rates of audio devices actually connected to a PC are 48 kHz and 44.1 kHz for music, 16 kHz and 8 kHz for voice, such as microphones and headphones, and 32 kHz, 24 kHz, 22.05 kHz, etc.

The bit rate of PCM (uncompressed sound source) with 16-bit quantization bits is as follows.

Stereo (for music) PCM 16-bit bit rate (example)
Sampling frequency Number of quantization bits Number of channels bit rate Comments
48 kHz 16 2 1,536 kbps
44.1 kHz 16 2 1,411.2 kbps Music CD
32 kHz 16 2 1,024 kbps
24 kHz 16 2 768 kbps
22.05 kHz 16 2 705.6 kbps

Methods of compression and compression of audio signals Part 3

Methods of compression and compression of audio signals Part 3

Audio Compression

The most popular compression format today is MP3.

The MP3 (MPEG Layer 3) format was developed, after several intermediate formats, by the Fraunhofer Institute in Germany. Actually, the .MP3 format relies on fooling the human ear. After some research, it turned out that human hearing tends to adapt to the appearance of new sounds, which is expressed in an increase in the hearing threshold. Therefore, some sounds are capable of masking (that is, making them subjectively inaudible) others. So in this format, some of the sounds that, according to the corresponding theory, are made inaudible, are simply removed from the general sound. The resulting “semi-finished product” is then encoded using the Hoffman method. Be sure to note that in the MP3 format, programs that compress the sound of the original are not standardized, that is, each competent programmer can implement their own compression scheme. And only the decoders obey the standards, which leads to the fact that the quality of MP3 playback does not always depend on the player that plays this file. Due to the different abilities and predilections of implementers of various encoders, some of them are better at handling symphonic music, some at rock and metal, some at rap and rave, etc.

JointStereo, which is one of the features of MP3, means that instead of encoding stereo as two independent channels, it encodes the call. center channel and the difference from the original stereo channels. Many stereo channel audio components are the same, and encoding them on the common channel allows you to free up additional bandwidth for more detailed encoding of the difference, leading to improved quality.

Be sure to mention the variable bit rate or VBR. This means that the encoder changes the compression ratio on the fly, depending on the nature of the sound. This approach results in a reduction in the final file size or, if quality requirements increase, the same file size produces better sound.

MP3 Pro – Introduced in 2001, the MP3 Pro codec was developed by Coding Technologies in association with Thomson Multimedia. It is MP3 based and as a result it turned out to be fully MP3 backward compatible and only partially forward compatible. It uses SBR (Spectral Band Replication) technology, so the codec provides good quality at low bit rates. However, the encoding quality at medium to high bit rates is inferior to almost all other codecs. As a result, MP3 Pro is used more for streaming on the Internet and demonstrating snippets of new musical compositions.

The MPEG-4 audio standard does not require a single or small set of highly efficient compression schemes, but rather a complex set to perform a wide range of operations, from low-quality speech coding to high-quality music and audio synthesis.

The MPEG-4 family of audio coding algorithms ranges from low quality voice (up to 2 kbps) to high quality audio (64 kbps per channel and higher).

RAW – Yes, it is not just the image format in which some digital cameras write photographs. In fact, RAW is the so-called. “Pure digitization”, which does not contain a title and contains only a sequence of samples of a sound wave. Typically, the scan is stored in 16-bit format.

Shorten is one of the first lossless codecs to appear. For a long time the project “slept sweetly.” However, in 2007, it began to develop again.

TTA (True Audio) – Finally about the most interesting. TTA is being developed by a team of our compatriots. And, I must say, the result of their work is impressive. All in order.

The codec is still quite young, but despite this it contains all the necessary features. We won’t list them again, we’ll just note that the format only lacks support for streaming audio over the network.

The format is open, as well as the source codes of the encoder program. There are compiled versions for Mac and Linux. There should be no compatibility issues during playback either, because there are already plugins for all popular players, as well as DirectShow filters for Windows Media Player. There is a plugin for Adobe Audition, which is important for musicians. For the past 4 years, hardware support has even appeared on players!

WAV – This is the primary audio format for many, many digital audio playback systems and is used as a standard audio file format on personal computers.

Compression and compression methods for audio signals Part 2

Compression and compression methods for audio signals Part 2

audio compression

FLAC is a member of the Xiph.Org codec family. By the way, it also includes the well-known ogg vorbis, one of the best lossy music compression algorithms. As a container for audio data, of course, OGG (files with the extension .ogg) and another open source container – Matroska (files with the extension .mka) are used.

It should be noted right away that both the FLAC format and algorithm are fully open. They are not patented, so they can be used completely free of charge in any program. This is the reason for the wide support for FLAC in players – any serious gamer has a plugin for FLAC. In addition, there are hardware mp3 players that support the FLAC codec.

The FLAC encoder is compiled for most platforms in use, so there should be no compatibility issues on alternative Windows operating systems.

FLAC supports tags in its own “FlacTags” format. There is the ability to encode multi-channel audio, a great advantage over Monkey’s Audio. The format supports any sample rate in the range of 1 Hz (!) To 65,535 Hz. Audio bit depth from 4 (!) To 32 bits.

FLAC is believed to be the most efficient use of system resources when decoding (playing) audio compared to other lossless codecs. Unfortunately, this is achieved at the expense of a significant increase in encoding (compression) time.

The FLAC website is regularly updated and new versions of the codec are released. Overall, FLAC is without a doubt the leader in terms of development activity. This may make it the main format in the future. Well, let’s see …

FLAC is the best option for storing high quality music.

MIDI (Musical Instrument Digital Interface) is a standard for hardware and software that allows you to play (and record) music by executing / recording special commands, as well as the format of the files that contain those commands. The playback device or program is called a MIDI synthesizer (sequencer) and is actually an automatic musical instrument.

Unlike other formats, it does not store the digitized sound, but sets of commands (played notes, links to played instruments, variable sound parameter values) that can be played differently depending on the playback device. The convenience of the MIDI format as a data representation format enables devices that produce automatic arrangements according to given chords, as well as 3D sound visualization applications. Additionally, these files tend to be orders of magnitude smaller than digitized audio of comparable quality.

Monkey’s Audio is a popular lossless digital audio encoding format. Distributed for free along with open source and a suite of encoding and playback software, as well as plugins for popular players. Monkey’s audio files use the following extensions: .ape to store audio and .apl to store metadata. Despite being open source, Monkey’s Audio is not free, as its license imposes significant restrictions on its use.

Audio files compressed with the Monkey audio codec have the extension ‘APE’; As you can see, the monkeys are present not only in the logo or the name (from English monkey: monkey, primate).

The average bit rate in an audio file is 600 to 700 kbps; compare with 128 kbps in MP3. Average compression is 40-50%, depending on the genre of music: if classical or jazz pieces are compressed in the best way, then compositions in the style of trash-metal or something similar “electronic noise” will show the worst result. . For codecs with acceptable quality loss, compression is approximately 80%.

There are four levels of compression. Maximum compression may seem like the only correct solution, although the compression time is quite long. However, you must also take into account the resource consumption of the system that plays the file; for the most compressed file, it is relatively high.

The .APE format provides tag support for searching for songs in your music collection. Another advantage is the verification of the integrity of the file during decoding. Recovery of original compressed .APE wav files is supported.

Monkey’s Audio has a graphical interface for Windows, in other words, a convenient window program to manage the encoding process. The rest of the codecs require the use of the command line or third-party interfaces.

Compression and compression methods of audio signals

Compression and compression methods of audio signals (types, differences, use)

Audio Compression

Basics of the analog-to-digital conversion principle, sound conversion and compression method, existing sound storage formats. Programs to convert and process sound and audio files. Application of these programs in linguistic research.

Bit rate is the amount of information per unit of time. In general, the bit rate is the number of bits that we spend encoding a sound with a duration of 1 second.

Analog-to-digital converter (ADC): A device that converts an input analog signal into a binary code (digital signal). The reverse conversion is done using a DAC (digital-to-analog converter, DAC). Typically, an ADC is an electronic device that converts voltage into a binary digital code. However, some non-electronic devices with digital output must also be classified as ADCs, such as some types of angle-to-code converters. The simplest one-bit binary ADC is a comparator.

The circuit to convert an audio signal from analog to digital:

Sampling is the transformation of continuous images and sound into a set of discrete values in the form of codes.

Quantization is the process of aligning a set of musical notes to a grid.

Compression (compression) of audio data is a process of lowering the bit rate by reducing the statistical and psychoacoustic redundancy of a digital audio signal.

The underlying idea behind all lossy audio compression techniques is to neglect the subtle details of the original sound that are beyond the reach of the human ear.

Codec (CoDec) is an abbreviation for compressor and decompressor. Basically, a codec is a collection of files, drivers, and libraries required to package a video or audio file into a compressed format and play the compressed file.

Formats:

AAC (Advanced Audio Coding) is an audio file format with less quality loss when encoding than MP3 of the same size. The format also allows you to compress without losing the quality of the source (ALAC AAC profile).

AAC (Advanced Audio Coding) was originally created as a successor to MP3 with improved encoding quality. The AAC format, officially known as ISO / IEC 13818-7, was released in 1997 as the new seventh part of the MPEG-2 family. There is also the AAC format known as MPEG-4

Apple AIFF: This file type is standard for Apple Macintosh systems and sound processing systems built on top of it. Apple AIFF stands for Audio Interchange File Format, an audio interchange file format, it is somewhat similar to WAV. Its peculiarity is that it allows you to place additional information along with the sound wave, in particular WaveTable samples (examples of the instrument sound together with synthesizer parameters), which improves the quality of the final result. Although today Apple computers are capable of playing files of almost any format, including MP3.

FLAC (Free Lossless Audio Codec) is a popular free codec for audio compression. Unlike lossy Ogg Vorbis, MP3 and AAC codecs, it does not remove any information from the audio stream and is suitable for both daily listening and archiving of audio collection. Today, the FLAC format is compatible with many audio applications.

Video with minimal loss of quality Part 2

Video with minimal loss of quality Part 2

There are special algorithms for filtering typical compressed video artifacts. Generally, any compressed video player includes post-processing of the decompressed video with such filters.

How do analog video standards work and on what principles are they based?
This question requires fundamental knowledge in the field of radio physics and engineering, and therefore a complete and detailed explanation is not included in the scope of this FAQ. In a significant simplification, the analog video signal encoding scheme of operation is as follows: the signal is transmitted in the form of timing and information components using partial quantization. The signal itself is analog, but it is divided (quantized) into lines and frames. The PAL standard (and close to it SECAM) assumes the presence of 25 frames per second, each of which contains 576 lines. The NTSC standard assumes 30 frames per second, each with 480 lines. The synchronization component carries information about the beginning / end of lines and frames. The information component carries the luminance component (Y), used in black and white television transmission, and two color components (U / V).

Capture questions
What is video capture?
Video capture is the process of converting analog video to digital format and then storing it on a digital storage medium. The most typical example of video capture is digitizing a TV broadcast or VHS cassette on a specially equipped PC.

What are the minimum system requirements for video capture?
First, you need a video capture device capable of delivering frames of a video stream at a sufficient speed in one of the standard color-coding formats. Such a device is a TV tuner or a video card equipped with a video input. There are also more expensive and “advanced” devices that can immediately compress captured video, but their consideration is beyond the scope of these FAQs.

Second, you will need enough hard disk how fast with a large capacity, capable of recording high bit rate video. So to capture video, which is then supposed to be recorded to a VideoCD (352×288 YUV12), the hard drive must be capable of recording 4MB / s (not MB!). And to capture video with the best quality analog TV broadcast (for example, from a separate DVD player via the S-Video interface, or just from the TV broadcast), you will need a disc with a recording speed over 19MB / sec.

Third, you need a really fast CPU. . Especially if, to reduce the bit rate, the video will be compressed directly during capture. The minimum CPU required is considered Pentium-II 300, however, it is difficult to achieve “comfortable” work with the captured video on any processor with a frequency of less than 550 MHz. If you use MPEG4 algorithms for compression, the best option is Athlon 800-900 (and higher) or Pentium 4, which show excellent performance in tasks of this type.

Fourth, you will need at least 128MB of RAM; otherwise, any instability in the compression rate will have a very detrimental effect on the quality of the captured clip.

In what file format are you capturing the video?
The file format depends on the software used. Today, the three most common formats are AVI, MPEG, and ASF.

AVI (Audio-Video Interleaved – a format with interleaved blocks of audio and video information) is most often used when capturing. It was developed by Microsoft to store and play videos as part of the Video API for Windows. By its structure, this format refers to “chunks”: as in WAV or MIDI, all information is packed into headers, called chunks – chunks. This allows you to record video sequences both uncompressed and with any type of compression in an AVI file. The main disadvantages of AVI are its inability to stream (stream video over networks) and the 2GB file size limitation. And if the inability to stream does not have a noticeable effect on the video capture,

The MPEG format (the name comes from the Motion Picture Experts Group, a group of specialists in compression of video sequences), unlike AVI, involves the use of one of two standard compression algorithms: MPEG1 and MPEG2. MPEG1 is used to compress video with a small frame size (less than 288 vertically) and a bit rate of about 1-2 Mbps, and MPEG2 is used for video with a large frame size (more than 288 vertically) and a bit rate of around 5-10 Mbps.

Video with minimal loss of quality

Video with minimal loss of quality

Why do I need video on my computer?

In fact, what does storing video information give us on a computer and not on a normal videotape? Basically, digital video storage and processing is attractive for two reasons: First, digital video is not “stuck” from multiple views; secondly, it is essential to obtain a copy of the video material, absolutely identical to the original. These two reasons are enough to save your home video digitally.

What is the principle behind implementing video on a computer?
Just like the rest of the video and the movie. The frames are successively replaced 25 times per second (for Pal / Secam), which our brain perceives as a continuous movement. From this principle, we conclude: long video sequences are huge. So let’s say we save a video with a frame size of 352×288, 24 bits per pixel. It takes approximately 435 MB to save at least one minute of video. This is the main problem with digital video: an incredible gluttony in terms of file size.

How to deal with huge sizes?
There are two main ways of dealing with it: by reducing the basic parameters of video capture and compression. The main parameters of video capture are: frame size, color coding and frame rate.

The term “bit rate” is often introduced. Average bitrate is the size of a video stream in bits, relative to its length in seconds. The unit of bit rate is 1 bit / s – 1 bit per second (1bps – 1 bit (s) per second). Since 1 bit / s is very small for digital video, Kilobits / s (Kbps) and Megabits / s (Mbps) are also entered. For the video in the previous question, the bit rate is 58 Mbps. The compressed video bit rate in VideoCD, which has the same frame size and frame rate, is 1.1 Mbps.

What are the main types of compression?
Compression is divided into two types: lossless (often called “lossless” for short) and lossy (“lossy”). The difference between these types is clearly apparent from their name. Most lossless compression methods do not take into account the visual similarity of adjacent frames in a video stream. In contrast, lossy compression techniques use this similarity in most cases. Because of this, the maximum compression ratio of an average video clip achieved by lossless algorithms does not exceed 3 to 1, while algorithms that operate with loss of quality can compress up to 100 to 1.

Very often, methods that take into account the similarity of adjacent frames in a video sequence are called “recursive.” Only individual frames are fully saved to them, called keyframes (sometimes intra). All other frames contain only differences from the previous ones (sometimes they also contain links to information contained in the following frame).

Techniques that compress each frame separately from the others are called “separate”. In video sequences compressed by such methods, all frames are keyframes.

Lossless compression often uses algorithms similar to those used in archive archives (ZIP, RAR, LZH). Algorithms that use the discrete Fourier transform with the preservation of such a number of coefficients, which is sufficient for the complete restoration of the original information, are compressed more strongly.

In lossy compression methods, algorithms based on the discrete Fourier transform are also used more frequently, but the number of stored coefficients is almost always much smaller than algorithms that work without loss of quality. Lossy compression generally encodes the amount of quality reduction for each frame or the average or instantaneous bit rate of the output file.

What are the problems with video compression?
Lead the so-called. artifacts. Artifacts are visible violations of the video stream quality that occur during the lossy compression process. The most common compression artifact is dividing an image into square blocks. “Recursive” compression algorithms also often have “garbage” near contrasting edges and moving objects, as well as “wavy” and “fuzzy” colors. The number and intensity of artifacts depend on the compression algorithm, the ratio of the output bit rate to the original, and the nature of the compressed image.

How to Compress Video “Lossless” (Reduce Video Size) Part 3

How to Compress Video “Lossless” (Reduce Video Size) Part 3

How much space does HD Ready, Full HD, 4K video take up?
On average, it takes 1 minute of video:

When shooting in HD Readry (720p) at 30 frames / second – 60 MB;
When recording in Full HD (1080p) at 30 frames / second – 130 MB;
When recording in Full HD (1080p) at 60 frames / second: 200 MB;
When recording 4K (UHD) at 30 frames / second: 375 MB.
The difference between HD, UHD, 4K and 8K video formats plays a decisive role in determining recording volumes. But there are other parameters that the video file size depends on: audio parameters, bit rate, aspect ratio, color resolution, and more.

Lossless video compression software
Hand brake
It is a free converter that converts video files of almost any format to mp4 (both DVD and Blu-Ray disc). Contains settings for popular tablets, androids, devices like iPhone, AppleTV, etc. Output file types:

video codec (H.265 (x265), H.264 (x264), MPEG-4, MPEG-2 (libav), Theora (libtheora), VP8 (libvpx);
Audio codec (Flac, AAC, CoreAudio AAC / HE-AAC (OS X only), MP3, Vorbis, or AC3)
audio tracks (DTS, AC-3, DTS-HD, MP3, AAC);
container (MKV and MP4 (M4V)).
For example: this program compresses HD video with mov format and sizes from three to four GB (duration 35-40 minutes) in mp4 format to 200-250 MB without losing its quality.

How to set up HandBrake correctly
Having launched HandBrake, you can see a lot of incomprehensible settings and what to do? Sooner or later you will realize, the main thing is to make an effort. So in this program you can edit videos and their sound.

Prepare your video for compression. Click the “Open Source” button and click the “File” item. Now we select our video.

The program contains blanks made by the developers, these are so-called presets. They mean ready-to-use parameters for any device or service. Very useful for those who do not want to dig deeper and understand the settings.

Each preset represents custom audio and video settings for iPhone, Android, Windows Phone, and other devices. Also, if suddenly this is not enough for you and you want to fine tune one or another parameter, after tuning it you can save your preset, which you can then use many more times.

Next, take a look at the most important features of Handbrake to manually configure the video conversion parameters.

In HandBrake, you can adjust the audio parameters. There is an “Audio” tab for this. In fact, audio can take up a lot of space in a video, especially if it is of good quality. In a simple video, so to speak, for the house, you can remove unnecessary tracks and leave only the first one. We configure “MP3” as codec. This will reduce the size of the sound while maintaining quality. We set the bit rate to 160, the sound quality will not be very good. If sound is important, set it to 320.

Choosing the best video container and codec
Ideally, when recording video, you should always set the maximum video settings for the device you are recording with. Only the maximum of a video camera will allow you to get the highest quality video, which can then be compressed with almost no loss of quality. The basis for video reduction is to use the most efficient codec and container for the video.

How to Compress Video “Without Loss of Quality” (Reduce Video Size) Part 2

How to Compress Video “Without Loss of Quality” (Reduce Video Size) Part 2

MPEG-2. There is the MPEG-2 codec (known as x.262) and there is also the MPEG-2 container format. MPEG-2 is used for over-the-air television signals and DVD video. In general, it provides an optimal quality / file size ratio. Supported by almost all players.

Microsoft VC-1. Microsoft VC-1 is used as an alternative to Adobe Flash technology and is used on the Microsoft Silverlight Internet platform. Provides medium compression quality.

MPEG-1. Obsolete video compression codec (replaced by MPEG2). Previously, such a video could be found on CD-ROM. Now it is less and less common, although if you like to watch old videos, you will probably come across it.

WMV. There is a Windows Media video codec and container with the same name. It provides very high quality video compression, with a pretty good picture. Very popular and widespread.

DivX is a video codec, the various versions of which comply with the MPEG-4 part 2, MPEG-4 part 10 (MPEG-4 AVC, H.264) standards.

Xvid (formerly “XviD”) is an MPEG-4 Part 2 video compression library. By the way, the compression methods used in MPEG-4 are proprietary, so it may be illegal to use Xvid in some countries.

About containers

There are now dozens of containers, each with its own pros and cons. The container, in addition to the video compressed with any codec, also contains sound (which is also compressed. Audio, as a rule, takes up a lot of space, so I do not consider audio codecs, their effect on the file size is not significant).

AVI (or Audio Video Interleave) is one of the most popular Microsoft containers. Most of the time, video on a PC is in this format. By the way, many, for some reason, do not recommend compressing modern videos in this format.

MP4. This container format was developed by Motion Pictures Expert Group. Many users call it MPEG-4. Usually video in MP4 files is encoded with the x.264 codec and audio is encoded with the AAC codec. Sometimes you can find other codecs in this container …

VOB and BDAV MPEG-2. It is used to package videos on DVD and Blu-ray discs. Blu-ray discs (.m2ts) can contain videos compressed with the x.264 and VC-1 codecs.

ASF (Advanced Systems Format) – Found in various extensions .asf, .wma, and .wmv. By the way, the .wmv file is most likely compressed with the WMV (Windows Media Video) codec, but the file itself is placed in an ASF container file.

QuickTime. The creation of Apple. The container supports many audio and video codecs. Apple is a supporter of x.264, therefore QuickTime files (formats: .mov, .qt) usually contain video compressed with the x.264 codec.

Flash. By the way, Adobe’s Flash container format is quite popular.

Matroska (.mkv, .mk3d, .mka, .mks) is a very popular and in-demand container today. Free, open source, that’s probably why it went mainstream. Many high-quality videos are distributed in this container.

DiVX. Both codecs and containers are very common. The format also has the same name: “.divx”. By the way, the codec has a “pirate” origin, which allows you to get high-quality videos with strong compression.

For a time, Divx was not recognized by the “big” equipment manufacturers, however, today many already incorporate the DiVX hardware codecs in most equipment (if it were not for the popularity of video files from this format, probably nothing would have happened …).

Which codec and container to choose

If you want maximum video compression, it makes sense to choose the x.264 codec, which provides excellent compression with good quality. However, keep in mind that not all computers (especially newer ones) allow you to view such files. If the video is very high resolution, even modern low-power PCs will not be able to play and display it without lag and crash. By the way, if the video is not compressed by any variation of the x.264 codec (for example, it is in MPEG1, 2 or some other format), it can be compressed without loss of quality into a more “compact” codec. (say, Divx).
If you plan to view the files on DVD players, various devices, etc. – it is recommended to choose the MPEG2 codec and container – it is compatible with almost all equipment;
For various movies, clips, cuts of the finished video, the MP4 container is excellent, as a good compromise between compression ratio and quality.