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What is an audio or video data container Part 2

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What is an audio or video data container Part 2

digital containers

First the proprietary DivX codec, then its open counterpart Xvid, helped burn a DVD to a CD. The quality was, of course, much worse than that of the DVD.
But an hour and a half of film took up 700 MB and in the 2000s the boom in film piracy was linked precisely to these codecs. If there were movies on the computer, they were movies of this format, with rare exceptions.
And since 2003 modernity begins. The Joint Video Team, under the auspices of the ubiquitous Moving Picture Experts Group, introduced the H.264 codec, which encodes video at the bottom of the post.

Well, almost after all, since then it was finalized, and YouTube generally outperformed my video in VP9 =) For example, in 2007, a plugin for H.264 was released – SVC (Scalable Video Coding), which not only complicated decoding and so on is not an easy codec for computers, it also made it possible to store video at various resolutions in such a format that the highest were dependent on the lowest. Most likely, you have seen pictures on the Internet in a progressive jeep, when they are not loaded from top to bottom, but first in squares, and then everything works out better until they are fully loaded. Here’s a similar story. With the advantage that devices that need to output video in a lower resolution than video, they may not waste resources decoding unnecessary layers.
And the codec consumes a lot of resources. It contains a lot of advanced technologies that I am unfortunately not strong at. However, today, even phones successfully cope with FullHD video in this format, and high-end ones also use 4K.

At the same time, the bit rate of such a 1080p video fluctuates around 2 Mbps, and even less without sound. And the fact of how much it is possible to reduce the amount of data by intelligently increasing the volume and complexity of the calculations still amazes me.

In 2006, Blurey records appeared.

In two years they have supplanted their competitor HD-DVD. They are still alive. The databases were developed by an entire consortium of large companies. Disks come in one layer and two layers, with capacities of 25 and 50 GB, respectively. The video for them is encoded in MPEG-2, MPEG-4, H.264 and the new Microsoft VC-1 codec at the time.
For HD-DVD, the capacities were a bit more modest (15 and 30 GB), but they could also be double-sided. The codec set is the same.

At the same time, the future is slowly approaching. Many would like to meet you in the person of the free VP9 codec, but it is most likely the corporate smile of H.265, which is also called HEVC. What can I say with the coming =)

Seriously, both codecs will find their place. Already today you can find video embeds on sites that are implemented in the open WebM format, which uses VP9 or 8. And since Google is only forcing the use of VP9, YouTube will also support both new codecs.
Both codecs are not revolutionary, but this is another round of video technology. Videos in H.264, VP8, H.265 and VP9 look great. Only the last two are smaller and have a higher usage ceiling. Another question is how much faster or slower will video be encoded in new formats, so that modest content producers like Slaylama are comfortable too. Yes, and these codecs have no special competitors, because today it is important again whether devices can decode video hardware to hardware – your smartphone can handle any open source Theora without any problem, but it will download much faster. So we have good and bad again, Coca-Cola and Pepsi, Android and iPhone, VP9 and H.265.

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What is an audio or video data container

What is an audio or video data container

Digital Container

A container is a file format or data stream that encodes data in one way or another.

Codecs and Containers

A codec is an encoder and decoder. Something that transforms the data. In the case of media, codecs are designed to compress the data stream and are often lossy.

Within the DV formats, the container can be AVI, Quicktime or the lesser known MXF. The codecs within these containers and formats may be different.

If we are talking about video compression, then there is a general rule of thumb: the more advanced the media we encode, the smaller the data stream or the file size, but more resources will be required for playback with subjectively equal recording quality.

Codec development took place in parallel with the growth of computer performance.

In 1988 the H.261 codec appeared. Few have heard of it, although it was in it that the concepts of keyframes, block vector transforms, and other technologies appeared, which are now used in all popular codecs.
That is, the video is not stored as a sequence of frames, as in a movie. The video is analyzed by the encoder, which finds an abrupt change in the image, for example the beginning of a new scene, and saves that frame, which is called the reference frame. And until the next keyframe, describe only the changes in this frame over time, dividing the image into blocks.

A group of MPEG-1 compression standards has been developed in the 93rd Moving Images Expert Group (MPEG), formed by the International Organization for Standardization (ISO).

With respect to H.261, it was possible to construct changes not only from the previous frame of reference, but also from the next one; and also to encode some section isolated from the rest.

In 1996 MPEG-2 appeared. It is for them that DVDs will be encoded later, you can imagine the scale of distribution. Interlaced scanning is back in the game, so there is nothing radically new.
It is necessary to dwell on the DVD video in detail. These discs appeared in the distant 96, and by 2003 they had become the leading consumer video format.
Movies were recorded at 720 × 576 pixels, which is the same as the D1 format. At the same time, compression made it possible to reduce the bit rate, that is, the data stream, to 9.8 Mbps, which made it possible to write movies on 4.7 GB discs. Coding format: 4: 2: 0, with decreased resolution of the chrominance channels; This trick allows you to reduce the size of the files without significantly affecting the quality of the image, because the luminance channel remains at the original resolution.
The third MPEG does not exist separately, all its chips were absorbed by the second. It also has nothing to do with mp3. They started developing it roughly on par with the second, aiming for higher bit rates, but then they solved all their tasks within the MPEG2 framework.

Understand the difference between encoding and container format.

Understand the difference between encoding and container format.

A container format is a data format that “encapsulates” other encoded data. It often contains “meta information” about the encoded data, or a way to store several separate streams of encoded data, or something like that.

The encoding produced by the codec is the real essence of the data stream.

The most common example I can think of is the Ogg / Vorbis format. Ogg is the container format and Vorbis is the encoding. So you have a file in Ogg format and inside it are these little buckets that contain encoded data. Inside each repository is a stream of Vorbis encoded data and nothing else. For example, in the cube, you can include the name of the artist and the title of the song.

So, back to technology:

If you already have lossy music like mp3 or ogg / vorbis, converting it to lossless format will only consume (a lot) of disk space, and W NOT NOT NOT NOT will improve the overall sound quality. You cannot create loyalty once it is lost. Unless you are writing a Visual Basic GUI in some successful program called CSI, but this is fantasy, not reality.

If you have music in other lossless formats and want to convert them to FLAC, you can.

Be careful when dismissing the term “WAV”. Wav doesn’t have to be lossless; In fact, WAV is just a container for the various possible formats. In this sense, it is similar to AVI. You can have lossless WAV if it is just raw PCM data, but you can also embed MPEG-1 (lossy) layer 3 data into a WAV file.

When converting from one lossless format to another, you may lose data if you reduce the accuracy of the data. For example, if you convert an unsigned 16-bit 48000 Hz data stream to an 8-bit 44100 Hz PCM data stream, you lose precision in two ways: samples are combined from 48000 to 44100 for every second (whichever results in data loss), and the data needs to be scrambled to match only 8 bits of information rather than 16 per sample, which will drastically degrade the quality.

Every digital audio stream, even those encoded with a lossy or lossless compression encoder, has the following sample format properties, which are important elements that describe the properties of the stream:

Sample width and bit depth, i.e. 8-bit, 16-bit, etc. Bit width and bit depth are slightly different from each other, and there are also little-endian / big-endian (which does not affect quality) and signed or unsigned (which also does not affect quality, but does affect the way the encoder / decoder handles the data). The main thing to remember is that “more bits the better”. So 32-bit is better than 16-bit, etc.

Frequency, also known as the sample rate. More is better because you have more “samples” of sound played per second. Imagine sliding your finger across a deck of cards and seeing the cards blur – this is how digital audio works. Each sample is one card, and if you have more cards flying per second, the sound is smoother. For example, you would really notice if you were only flipping 5 cards per second, but everything would go blurry if you flip thousands of cards per second. So it’s even better, because it’s more natural and closer to reality, which is analogous and infinitely divisible (well, down to Planck units, but this is a moot point and off-topic).

Lossless simply means that if you use the same or better sample format in the output that you used in the input, you won’t lose any data.

So if you go from 16-bit to 32-bit format, you don’t lose data. But if you go from 32 bit to 16 bit, you lose data.

So the answer to your question about whether FLAC makes sense depends on the original data: if you have 64-bit WAV files that were originally recorded in this format, with 192,000 Hz (aka 192 kHz) and you convert them to “FLAC format 16-bit 44.1 kHz standard, you will lose TON of data. But if your WAV file is 8-bit with 22100 samples per second and you convert it to 16-bit FLAC with 44100 samples in second, you won’t lose data and even You can increase the file size depending on whether you gain lossless compression or a smaller sample format.

The sample format affects the amount of space the file takes up, so “bigger” bits and a “faster” sample rate take up more space.