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基于dsp的語音信號采集與回放系統(tǒng)的設計開題報告(完整版)

2025-04-30 12:42上一頁面

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【正文】 ream traffic are carried on the same wire and are presented as waveform. On digital systems, upstream and downstream traffic are carried on separate wires, but are synchronized to prevent interruptions within the call). In the IP world, the two end points are not fixed and are viewed as connectionless. Media RTP packets carrying voice data for a single call can be routed through different paths. As a result, packets of voice data arrive at the end point at different times (jitter) and out of sequence.To pensate for jitter, IP data networks use buffers to store ining packets. This gives delayed packets time to ‘catch up’ before the data is eventually sorted and passed to the end user. This system is designed for networks where real time guarantees are not required and delays in packet delivery is acceptable. However, on a telephone network delayed packets reduce voice quality. Packet buffering, though required on a VoIP network, must meet or exceed the standards of a telephone network (a maximum delay of 500 ms).Consider the following scenario: An ethernet cable is tapped for voice packets and the VoIP recorder intercepts the packets before they have been buffered. As a result, the packets pulled off the network are misaligned and predictably, the audio quality is poor. To pensate for this, hardware ponents used for VoIP call recording properly time the buffering of ining packets.Packet FilteringIn a conventional circuitbased telephone network, the line is used to transmit only voice data. On an IP network many types of packets data, voice and media are present on the same ethernet cable. Packet filtering is the selective passing or blocking of packets as they pass through a network interface. Packet filtering is used by VoIP recording systems to isolate voice related packets from data and media packets.Many earlystage VoIP recorders rely on host resources for packet filtering. This is a viable solution on networks with light traffic. However, this system is not scalable and quickly reaches limitation when the system density grows beyond 100 ports. The better solution is a logging system that uses hardware ponents capable of packet filtering. This system would no longer be limited by host resources and would provide a scalable solution for either low to high density environments.Voice CODECSAn important consideration in the design any logging system is its ability to encode/decode numerous pression schemes. Like all recording environments, the type of CODEC used for media transport is controlled by the network. As a result, when selecting hardware ponents for call recording, application developers prefer products that support multiple CODECs. This is crucial when tapping a VoIP network. When call setup is n
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