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Video codecs and containers.

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Video codecs and containers.

Video Codec

This article is intended to refer here to those who are trying to “convert” something, without understanding what they are doing and why.

Video Codecs

To work as efficiently as possible with any object, you need to understand how it works. If the video file is for you a mysterious black box, inside which mysterious things happen, perhaps not without the help of black magic, then your effectiveness will be minimal.

So. All information on the computer is in the form of files. This, I hope, is not a surprise to anyone. Here we will start from this basic concept.

Any video file must be a container. A container is a repository of content. There are multi-structure storages – these are container formats. For example, a bento box is an example of a container. You can put sushi or tempura on it. What can you put in a video container? Well, at least image and sound, one at a time. This is a set without which there is nothing to do. What can you put to the maximum? The modern Matryoshka container allows you to put various video and audio tracks, text and graphic subtitles, fonts to display them, images and I don’t know what else.

Going back to the bento box example, note that miso cannot be poured into it; will flow in fig. Not all containers can accept all flows. There are compatibility restrictions that make life difficult.

Container examples: mpeg, avi, mkv, mp4, ogm, vob, mov, rm, divx, asf. You don’t have to look closely at the list to understand that these are standard file extensions. Of course. Because file = container.

Streams or tracks are stored inside the container. These streams have a format called a codec. And this difference must be understood with particular clarity. The container is a file format. And the codec is the stream format it contains. They are two independent things. Yes, there are some inextricably linked containers and codecs. For example, the Real Media container can only store real video and real audio streams. And vice versa, these formats cannot be stored in any other container (almost, as I have already been corrected). But they are still different concepts that should not be confused.

The codec concept usually includes the following aspects:
1) The actual data storage format.
2) Software that allows you to encode information in this format and / or decode it from it.

Examples of video codecs: divx, xvid, avc, x264, vp6, vp7, mpeg-1, mpeg-2, huffyuv.
Examples of audio codecs: mp3, ogg, ac3, aac.

While containers are generally distinguished by file extensions, codecs are distinguished by the four-character FourCC code.

The codec concept is usually associated with a kind of compression. Raw (uncompressed) streams also have their own formats, but they do not require decoding, and therefore the concept of codec is generally not applied to them.

Now let’s take a look at the most popular containers, codecs, and related issues. As a general rule, the problems we have are of two types: related to reproduction and related to editing.

MPEG is one of the oldest containers. It can store only video in mpeg-1 format and audio in mp2 format. And in a friendly way, with quite strict restrictions on the size of the image and the bitrate of the sound. Due to the age and primitiveness of the format, almost all players and publishers understand it. But for the same reasons, it became almost impossible to meet him. Nobody needs these things.

AVI is also quite old, but it is still a very useful container. It’s good because, again, all the players and all the editors get it. Almost all mpeg-based formats fit into it, as well as many that support them. The following video formats do not fit avi: avc (aka Nero AVC or Nero H.264), wmv below version 9, as well as any tinsel like actual video, which was originally designed to be incompatible with anything in the world. By sounds, supposedly anything, except Vorbis ogg.

OGM is where Vorbis ogg goes. Because the format was created on the basis of this very ogg. At the moment, he is practically ousted by the matryoshka because he can do the same, only better. It is also not compatible with any conventional software.

MKV is a nesting doll that can fit just about anything except flash video. But due to its complexity and versatility, it is still possible to do with it only things like: mount, look and dismount.

MP4 is actually modern MPEG. It only takes things that are compatible with the MPEG standard, but at the same time includes its latest updates.

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Compressed audio encoding formats.

Compressed audio encoding formats.

audio encoding

MP3 (or rather, MPEG 1 Audio Level 3): no comment, compatible everywhere and by everyone, the lack of this “eternal” format is one: only two channels, which limits its use in cinema systems at home modern.
Multi-channel MP3 (5.1) MPEG 2 Audio Level 3.

audio encoding
WMA: Windows Media Audio, formally a better and more modern competitor to Microsoft’s mp3. It is not used much, although it is widely compatible with hardware.
OGG Vorbis is a best modern mp3 competitor from the open source community. Deprived of any license restrictions, it is used more and more frequently.
AAC: Advanced Audio Coding is Apple’s main audio format built into all of its iPads, iPhones, iTunes, etc. The main advantage is that it is technically more advanced than mp3, allowing sample rates of up to 96 kHz and theoretically a completely insane number of channels in one file, up to 48. It is also used in digital satellite radio. Just as mp3 is a compressed format, the quality of 96Kbps AAC is comparable to the quality of 128Kbps of mp3 (we are talking about two channels in both cases).
Dolby Digital (AC-3) is probably the most popular standard for digital audio in cinematography, due to the fact that it appeared on the market as early as 1995, it exists in two versions: DD2.0 (for high-quality stereo sound) and DD5 .1 – five full channels and one defective for a subwoofer. Players are compatible with all of them for obvious reasons, the bitrate is 640Kbps in all cases.
Dolby Digital Plus or E-AC-3 is an attempt to improve on the usual Dolby Digital, but the previous generation decoders and receivers do not support tracks in the Dolby Digital Plus format, the reasons for this are radical changes: the number of channels increased to 7.1, the bit rate – to 1, 7 Mbps This will not go through S / PDIF (when transmitting via such a cable, you will have to use downmix on DD5.1 or on DTS with quality loss), but HDMI normally copes with Dolby Digital Plus as of version 1.3, you can find such tracks on Blu-Ray discs …
Dolby TrueHD – We practically have 8 tracks almost uncompressed at 96 KHz / 24 bits or 6 at 192 KHz / 24 bits, the total bit rate reaches 18 Mbit / sec, which requires decoding in the player and transmission to the receiver in the analog path, or using HDMI 1.3 or higher. For Blu-Ray, this audio coding system is optional.
DTS is a lossy digital audio coding system for cinemas, which later appeared on DVD, it is analogous to Dolby Digital 5.1, but somewhat more flexible, allowing in addition to 2.0 and 5.1 to use other schemes, such as 4.0 and 4.1, there is also a choice between two fixed bit rates of 1500 Kbps and 750 Kbps. In the first case, DTS clearly outperforms Dolby Digital in sound quality; in the second, the difference between systems is controversial.
DTS-HD is a further evolution of DTS, the number of channels has been brought to 7.1 in 96KHz / 24bit mode, the bit rate can be selected between 6Mbps and 3Mbps, it is an optional audio format for Blu-Ray. The situation with the sound transmission to the receiver is almost the same as with DolbyTrueHD.

Lossless or uncompressed compressed audio encoding formats.

LPCM is simply uncompressed audio. It is usually stereo. It should not be confused with a WAV file, it is a container and there may be something other than PCM WAV inside.
APE is a specific lossless audio compression format. Loved by audiophiles.
Flac is its competitor and analog, the differences between them are beyond the scope of this review.
Lossless audio
Lossless apple

Subtitle formats.
SRT: text format, can be attached as a separate file with the same extension. Compared to the first versions of this format, the design possibilities have been significantly increased. It can also exist within MKV.
SUB / IDX is a graphic subtitle format extracted from DVD. It can fit MKV or MP4.
s2k, ssa, ass: some more advanced text formats, ass can be placed inside MKV.
smi is a textual format based on SGML, the direct ancestor of HTML.
PGS is a graphical subtitle format, the main one for Blu-Ray, but it can also exist in ts and MKV containers.

Audio encoding and processing

Audio encoding and processing

Audio processing

Sound information. Sound is a wave that travels through air, water, or other medium with a continuously changing intensity and frequency.

Audio processing

A person perceives sound waves (air vibrations) with the help of hearing in the form of sound of different volume and pitch. The higher the intensity of the sound wave, the louder the sound, the higher the frequency of the wave, the higher the pitch of the sound

The human ear perceives sound at a frequency of 20 vibrations per second (low sound) to 20,000 vibrations per second (high sound).

A person can perceive sound in a wide range of intensities, in which the maximum intensity is 10 14 times greater than the minimum (one hundred thousand billion times). To measure the volume of sound, a special unit “decibel” (dbl) is used (Table 5.1). Decreasing or increasing the sound volume by 10 dB corresponds to a decrease or increase in sound intensity by 10 times.

Table 5.1. Sound volume
Sound Volume in decibels
Lower limit of human ear sensitivity 0
Leaf whisper ten
Conversation 60
Horn 90
Jet engine 120
Pain threshold 140
Sound time sampling. In order for a computer to process sound, a continuous audio signal must be converted to a discrete digital form using time sampling. A continuous sound wave is divided into separate small time sections, for each section a certain value of sound intensity is set.

Therefore, the continuous dependence of the loudness of the sound at time A (t) is replaced by a discrete sequence of loudness levels. On the graph, this appears to replace a smooth curve with a sequence of “steps”

Sampling frequency. A microphone connected to the sound card is used to record analog sound and convert it to digital format. The quality of the digital sound obtained depends on the number of measurements of the sound volume level per unit of time, that is, the sampling frequency. The more measurements that are made in 1 second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal repeats the curve of the dialogue signal.

Audio sample rate is the number of audio volume measurements in one second.

The audio sample rate can vary between 8000 and 48000 sound volume measurements per second.

Audio encoding depth. Each “step” is assigned a specific value for the sound volume level. Loudness levels of sound can be viewed as a set of possible states N, for which a certain amount of information I is required, which is called audio coding depth.

Audio encoding depth is the amount of information required to encode the discrete volume levels of digital audio.

If the known encoding depth, the number of digital audio volume levels can be calculated using the formula N = 2 I. Let the audio encoding depth be 16 bit, then the number of sound volume levels is:

N = 2 I = 2 16 = 65 536.

During the encoding process, each sound volume level is assigned its own 16-bit binary code, the lowest sound level will correspond to the code 0000000000000000 and the highest – 1111111111111111.

The quality of digitized sound. The higher the sampling frequency and depth of the sound, the better the sound of the digitized sound. The lowest quality of digitized sound, corresponding to the quality of telephone communication, is obtained at a sampling rate of 8000 times per second, a sampling rate of 8 bits, and by recording an audio track (“mono” mode). The highest quality of digitized sound, corresponding to the quality of an audio CD, is achieved with a sampling rate of 48,000 times per second, a sampling rate of 16 bits and the recording of two audio tracks (stereo mode) .

It should be remembered that the higher the quality of the digital sound, the greater the volume of information in the audio file. It is possible to estimate the volume of information of a digital stereo sound file with a duration of 1 second with an average sound quality (16 bits, 24,000 measurements per second). To do this, the encoding depth must be multiplied by the number of measurements in 1 second and multiplied by 2 (stereo sound):

16 bits × 24,000 × 2 = 768,000 bits = 96,000 bytes = 93.75 KB.

Sound editors. Sound editors allow you not only to record and play sound, but also to edit it.

Audio encoding: secrets revealed

Audio encoding: secrets revealed

audio encoding

Audio settings for video capture and transmission.
As people directly connected to the AV sphere, we constantly talk about audio coding and audio codecs, but what is it? An audio codec is essentially a device or algorithm that can encode and decode a digital audio signal.

Audio Encoding

In practice, the audio waves that are transmitted over the air are continuous analog signals. Signals are converted to digital format by a device called an analog-to-digital converter (ADC), and the reverse conversion device is a digital-to-analog converter (DAC). The codec is between these two functions and it is he who allows you to adjust some important parameters for the successful capture, recording and transmission of an audio signal: codec algorithm, sample rate, bit depth and data rate.

The three most popular audio codecs are Pulse-Code Modulation (PCM), MP3, and Advanced Audio Coding (AAC). The choice of codec determines the compression rate and the recording quality. PCM is a codec used by computers, CDs, digital phones, and sometimes SACD. The source of the PCM signal is sampled at regular intervals and each sample is the digital amplitude of the analog signal. PCM is the simplest option for digitizing an analog signal.

With the correct parameters, this digitized signal can be completely converted back to analog without any loss. Unfortunately, this codec, which provides almost complete identity with the original audio, is not very cheap, which results in large files, and these files are not suitable for streaming. We recommend using PCM to record digital images for your sources or when doing audio post-processing.

Fortunately, we always have the option of choosing a different codec that can compress digital data (rather than PCM) based on some helpful observations on the behavior of sound waves. But in this case, you have to make a compromise: all alternative algorithms are associated with “losses”, since it is impossible to completely restore the original signal, but nevertheless the result is so good that most users will not be able to notice the difference.

MP3 is an audio encoding format that uses a digital data compression algorithm that allows you to save the audio signal in smaller files. The MP3 codec is the most used by users to record and store music files. We recommend using MP3 to stream audio content as it requires less network bandwidth.

AAC is a newer audio encoding algorithm that is the successor to MP3. AAC has become the standard for MPEG-2 and MPEG-4 formats. In fact, this is also a digital data compression codec, but with less quality loss than MP3 when encoded with the same bit rate. We recommend using this codec for online streaming.

Sampling frequency (kHz, kHz)
Sample rate (or sample rate): the frequency with which the signal is digitized, stored, processed or converted from analog to digital. Time sampling means that the signal is represented by a number of its samples (samples) taken at regular intervals.

Measured in hertz (Hz, Hz) or kilohertz (kHz, kHz,) 1 kHz equals 1000 Hz. For example, 44,100 samples per second can be labeled 44,100 Hz or 44.1 kHz. The selected sample rate will determine the maximum playback frequency and, as follows from Kotelnikov’s theorem, to fully restore the original signal, the sample rate must be twice the highest frequency in the signal spectrum.

As you know, the human ear is capable of picking up frequencies between 20 Hz and 20 kHz. Given these parameters and the values shown in the table below, you can understand why 44.1 kHz was chosen as the sampling frequency for CD and is still considered a very good frequency for recording.

There are several reasons for choosing a higher sample rate, although it may seem like a waste of time and effort to reproduce sound outside the range of human hearing. At the same time, 44.1 – 48 kHz will suffice for the average listener for a high-quality solution to most problems.

Bit depth
Along with the sample rate, there is the bit depth or depth of the sound. Bit depth is the number of bits of digital information to encode each sample. Simply put, bit depth determines the “accuracy” of the input signal measurement. The larger the digit capacity, the smaller the error for each individual conversion from the magnitude of an electrical signal to a number and vice versa.