Considering AVS as video and AAC as audio for transmission of digital multimedia content over a broadcast network provides the end users to take advantages of such leading technologies. In order for a proper transmission of multimedia content, the video and audio streams cannot be transmitted separately. These streams need to be multiplexed before transmission. The objective of the thesis is to propose an effective method for multiplexing the AVS video and AAC audio elementary streams and is followed by de-multiplexing the stream at the receiver end while achieving lip synchronization between the audio and video streams during playback. Since the video and audio streams have a frame wise arrangement, frame numbers are used as synchronization information during multiplexing which helps in achieving lip synchronization between the video and audio streams during playback. There are totally two layers of packetization adopted for multiplexing the video and audio streams before their transmission. The synchronization information is embedded in the headers of the first layer of packetization. The packetization layers adopted in the thesis conform to the MPEG-2 systems standard [25], which meets the various requirements of the transmission channels. In order to prevent the buffer under or over flow at the receiver end, play back time has been chosen as reference criteria by which the audio and video data packets are allocated according to their play back time in the multiplexed stream. The advantages and limitations of the proposed method are discussed in detail.
Considering AVS as video and AAC as audio for transmission of digital multimedia content over a broadcast network provides the end users to take advantages of such leading technologies. In order for a proper transmission of multimedia content, the video and audio streams cannot be transmitted separately. These streams need to be multiplexed before transmission. The objective of the thesis is to propose an effective method for multiplexing the AVS video and AAC audio elementary streams and is followed by de-multiplexing the stream at the receiver end while achieving lip synchronization between the audio and video streams during playback. Since the video and audio streams have a frame wise arrangement, frame numbers are used as synchronization information during multiplexing which helps in achieving lip synchronization between the video and audio streams during playback. There are totally two layers of packetization adopted for multiplexing the video and audio streams before their transmission. The synchronization information is embedded in the headers of the first layer of packetization. The packetization layers adopted in the thesis conform to the MPEG-2 systems standard [25], which meets the various requirements of the transmission channels. In order to prevent the buffer under or over flow at the receiver end, play back time has been chosen as reference criteria by which the audio and video data packets are allocated according to their play back time in the multiplexed stream. The advantages and limitations of the proposed method are discussed in detail.
swamithesis.docx