Introduction to Digital Video and Sound Transmission Principles

First, the concept of code stream

An important concept involved in digital video and sound transmission is the so-called "code stream" concept. The so-called streaming media refers to the simultaneous transmission of video, sound and data from the source to the destination, it can be received as a continuous real-time stream at the destination. The source here refers to the server-side application, while the destination or receiver refers to the client application.

After the streaming data is transmitted from the server-side application, it can be received and displayed or played back by the client application. Generally, after the client application receives enough data and stores it in the buffer, it immediately displays the video or plays back the audio.

An important feature of streaming media is its sensitivity to time, which is necessary for applications with high real-time requirements, so it is natural that such applications are inseparable from streaming media. The realization of streaming media mainly depends on the improvement of network bandwidth and compression algorithm. Today, with the improvement of network protocols, network infrastructure and compression technology, the realization of streaming media has become easier.

2. Stream transmission method

There are three main transmission technologies for streaming media: point-to-point (unicast), multicast (Multicast) and broadcast (Broadcast). Multicast is also called multicast. The characteristic of point-to-point is that the source and destination of streaming media are one-to-one correspondence, that is, streaming media can only reach one destination (client application) after being sent from a source (application on the server side). Multicast is a "group" -based broadcast, and its source and destination are in a one-to-many relationship, but this one-to-many relationship can only be established in the same group, that is, streaming media from a source (Server-side application) After sending it, any destination (client application) that has been added to the same group number as the source can receive it, but other destinations (client application) outside the group cannot receive it. . The source and destination of the broadcast is also a one-to-many relationship, but this one-to-many relationship is not limited to groups, that is, after the code stream is sent from a source (server-side application), the same network segment All destinations (client applications) can be received, and broadcast can be regarded as a special case of multicast.
Broadcast and multicast are very meaningful for streaming media transmission, because the data volume of streaming media is often very large and requires a large amount of network bandwidth. If the point-to-point method is used, as many destinations as possible have to transmit as many streaming media, so the required network bandwidth is proportional to the number of destinations. If broadcast or multicast is used, then the streaming media only needs to be transmitted at the source One copy can be received by all client applications in the group or on the same network segment, which greatly reduces the occupation of network bandwidth.

3. Digital video and sound transmission technology

Digital video and sound transmission belong to the category of streaming media transmission. After the analog video and sound signals are converted into digital form by the capture device, the data volume is very amazing. If compression technology is not used, it is unthinkable to realize the network transmission of digital video and sound. On the other hand, digital video and sound transmission are very sensitive to time and have high real-time requirements. It is difficult to meet the requirements without using a special network transmission protocol. Therefore, the general approach to achieve digital video and sound transmission is to compress the digital video and sound information at the source end, and then transmit it to the destination via a network such as ATM with quality of service (ie QoS) guarantee, and then at the destination Decompress it and display or play it back. If you need to transmit on a network such as an IP network that does not have QoS guarantees, you must at least use Real-time Transport Protocol (RTP) for transmission.

There are many types of digital video and audio compression technologies that have been developed and are being developed. Different compression technologies have different emphases and are adapted to different applications. Some of these compression technologies have been standardized, but many are not. Commonly used standardized compression technologies are MPEG-1, MPEG-2, H.261 / H.263, etc., and MPEG-4 is being developed. MPEG-1, MPEG-2 are suitable for high-bandwidth video and audio applications that can provide high quality and low latency, while H.261, H.263 and MPEG-4 are being developed for low-bandwidth image quality delay Undemanding applications.

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