Design plan for video call based on STi7105 HD set-top box

Abstract: This article is based on the STI7105 high-definition set-top box and uses the SIP protocol to use the existing codec capabilities of the high-definition interactive set-top box as much as possible to realize the video call function. This solution makes full use of the existing modules of the set-top box and the HFC network, but the compression of voice and video is not high.

1 Introduction

With the development of science and technology, especially multimedia communication technology, the traditional voice telephone model is increasingly unable to meet the needs of modern people for information exchange. As multimedia data such as images, audio, and video have gradually become the main media expression in the field of information processing, compared with traditional voice telephony, video telephony can attract users more, and can effectively use network resources, greatly reducing costs. The cable TV network has already realized video and data services in business. In order to adapt to the pace of China's vigorous promotion of "triple network integration", it must be able to realize communication services, so video calling is a good choice.

2 STI7105 chip

STI7105 is a high-performance high-definition digital TV set-top box decoder chip produced by STMicroelectronics. It adopts advanced semiconductor production technology and perfect optimized design to improve the performance of each functional module, increase high-speed data interface, reduce power consumption, and reduce system The overall material cost. STI7105 is a SoC chip with a wide range of applications, which can be used for cable high-definition digital TV set-top boxes, satellite high-definition digital TV set-top boxes, IPTV set-top boxes, multimedia handheld devices, etc. The structure diagram of STi7105 chip is shown in Figure 1.

Figure 1 STi7105 schematic diagram

Figure 1 STi7105 schematic diagram

The STi7105 chip integrates a high-performance ST40 application processor with a main frequency of up to 450MHz and real-time processing and calculation capabilities exceeding 800DMIPS. In addition, two real-time calculation accelerators are integrated in the chip, which can also be used for various processing requirements . This processing capability can not only meet the requirements of high-definition set-top boxes, but also fully meet the design requirements of home multimedia centers.

STi7105's TS stream input processing module can support 4 TS stream input and 1 TS stream output. This processing module can simultaneously process and decompose multiple code streams for simultaneous recording, code stream playback, picture-in-picture and other functions. These streams can come from traditional digital TV broadcast channels, or from IP channels such as Cable Modem, EoC, Ethernet, etc. This module can provide users with flexible product development modes.

STi7105 integrates a high-performance video and audio decoder. Its flexible decoding architecture can meet the decoding requirements of various existing video and audio encoding and compression formats. It can also support these standards through software upgrades. In terms of video encoding and decoding, it can support H.264HP@HL4.1, MPEG-2 MP @ HL, VC-1 MP @ HL, AP @ L3, MPEG-4 P2 ASP @ L5, DiVx format, RM, RMVB, AVS and other video formats, the decoder can be used for PVR, picture-in-picture functions, support 1080p format, built-in image noise reduction processing, enhanced optimized post-processing circuit can bring users unprecedented beautiful images. In terms of audio codec, it can support MPEG-1 layer I / II, MP3, MPEG-2 layer II, AC-3 Dolby Digital, DD +, AAC, AAC + SBR, etc., as well as Dolby, DTS new algorithms And China DRA audio standard.

In addition to the video and audio decoding function, STi7105 also has certain video and audio encoding capabilities, and supports DivX, XviD, and H.263 encoding formats.

3 Implementation and application

3.1 Technical Agreement

The development and application of video call terminal software involves many aspects of technology, including signaling protocols, packet voice technology, video codec technology, and streaming media network transmission technology.

3.1.1 Signaling protocol

Signaling refers to a communication language used by various exchanges to complete call connection. Various control signals passed between communication devices, such as occupancy, release, device busy status, and called user number, are all signaling. The signaling system guides the various parts of the system to cooperate with each other and work together to complete a certain task. At present, the main international standards for IP network communication are H.323 and SIP (Session Initiation Protocol), both of which provide a complete solution for video telephony system signaling, both of which are used as application layer control (signaling) for multimedia communication. The protocols all use RTP as the media transmission protocol. But the design styles of the two are quite different. H.323 uses the traditional telephone signaling mode, including a series of protocols; while SIP draws on other Internet protocols and uses text-based protocols.

The SIP protocol is increasingly favored by the communications industry because of its simplicity, easy expansion, and ease of implementation. SIP is very important in the evolution of existing communication networks to NGN, and is becoming one of the core protocols of NGN (Next Generation Network) And, the SIP protocol has been defined by the 3GPP working group as the signaling protocol of the third-generation mobile communication system to provide IP multimedia services. This solution uses SIP as the signaling protocol.

The SIP protocol is formulated by the IETF for multi-party multimedia communication. The main purpose is to solve the signaling control in the IP network and the communication with the soft switch, thereby forming a new generation communication platform. The SIP protocol uses a client / server working method. The SIP network contains two types of components: a user agent and a network server.

User agent (UA) represents a terminal system, including a user agent client (UAC) and a user agent server (UAS), the former generates a request and the latter generates a corresponding response. UAC and UAS are two logical parts, and each end system includes the functions of UAC and UAS.

SIP is a protocol with a hierarchical architecture. Layer 1 is the syntax and coding layer, and its coding uses the extended Backus paradigm. Layer 2 is the transport layer, which defines how clients and servers on the network receive requests. And send responses; the third layer is the transaction layer. The transaction is the basic element of SIP. The transaction is composed of the request sent by the client to the server and all responses sent from the server to the client. If the transaction layer times out, the transaction layer is retransmitted. Match the response to the request; the third layer is the user layer. Each SIP entity (except the stateless proxy) is a transaction user. When a transaction user wishes to send a request, a client transaction instance is created to send the request. SIP provides functions such as user positioning, user communication capability negotiation, user availability, call establishment, and call processing and control. SIP does not provide specific services, does not define how the session will be described, only provides the session establishment function. The SIP protocol clearly distinguishes between session establishment and session description, so that SIP can become a signaling protocol in the real sense of multimedia sessions on the Internet, and has a wide range of applications.

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