Concept of code stream
An important concept involved in digital video and sound transmission is the so-called "code stream" concept. Streaming media refers to the transmission of videos, sounds, and data from the source to the destination at the same time. It can be received as a continuous real-time stream at the destination. Here, the source refers to the server-side application, and the destination or acceptor refers to the client application.
After the stream data is transmitted from the server application, it can be received and displayed or played back by the client application, generally, the client application receives enough data and stores it in the buffer zone to immediately display the video or play back the audio.
One important feature of streaming media is its sensitivity to time, which is essential for applications with high real-time requirements. Therefore, such applications are inseparable from streaming media. The implementation of streaming media mainly depends on the improvement of network bandwidth and compression algorithm. Today, with the improvement of network protocols, the development of network infrastructure and compression technology, the realization of streaming media has become more and more easy.
Ii. code stream transmission mode
Streaming media transmission technologies include point-to-point (unicast), multi-access broadcast (Multicast), and broadcast (broadcast ). Multicast is also called multicast. Point-to-point feature is that the source and destination of streaming media are one-to-one, that is, streaming media can only reach one destination (client application) after being sent from one source (server application ). Multicast is a broadcast based on "group". Its source and destination are one-to-many relationships. However, this one-to-many relationship can only be established in the same group. That is to say, after the streaming media is sent from a source (server-side application), any destination (client application) that has been added with the same group number as the source can receive it, however, none of the other destinations (Client Applications) in this group can receive them. The source and destination of the broadcast are also one-to-many relationships, but these one-to-many relationships are not limited to groups. That is to say, after the stream is sent from a source (server-side application, all the destinations (Client Applications) on the same network segment can be received. Broadcast can be seen as a special case of multicast.
Broadcast and multicast are meaningful for streaming media transmission, because the data volume of streaming media is usually large and requires a large amount of network bandwidth. If you use the point-to-point method, You have to transmit as many streaming media as the number of destinations. Therefore, the required network bandwidth is proportional to the number of destinations. If you use the broadcast or multicast method, streaming media only needs to transmit one copy at the source end, and all client applications in the group or on the same network segment can receive it, which greatly reduces the usage of network bandwidth.
Iii. digital video and sound transmission technology
Digital video and sound transmission belong to the category of streaming media transmission. After analog video and sound signals are converted to digital form by capturing devices, the data volume is amazing. If compression technology is not used, it is unimaginable to achieve network transmission of digital videos and sounds. On the other hand, digital video and sound transmission are highly time-sensitive and have high real-time requirements. If special network transmission protocols are not used, it is difficult to meet the requirements. Therefore, the general practice of digital video and sound transmission is to first compress the digital video and sound information at the source end, and then use QoS such as ATM) ensure that the network is transmitted to the destination, and then decompress it at the destination to display or play back it out. If you want to transmit data over a network without QoS guarantee, such as an IP address, you must at least use the real-time transmission protocol (RTP) for transmission.
At present, there are many developing digital video and audio compression technologies. Different compression technologies have different focuses to adapt to different applications. Some of these compression technologies have been standardized, but many have not yet been standardized. Commonly used has been standardized compression technology MPEG-1, MPEG-2, H.261/H.263, etc., is developing MPEG-4 and so on. MPEG-1 and MPEG-2 for high-bandwidth video and audio applications that provide high quality and low latency, while H.261, H.263 and developing MPEG-4 are suitable for low-bandwidth applications with low image quality latency requirements.