以超高清(Ultra High Definition,UHD)视频技术及应用为研究重点,首先全面介绍超高清视频技术的概述,其次重点分析超高清视频产业链关键组成部分,最后深入探讨超高清视频技术的应用领域,包括在电影、电视、文教娱乐等领域的应用,为超高...以超高清(Ultra High Definition,UHD)视频技术及应用为研究重点,首先全面介绍超高清视频技术的概述,其次重点分析超高清视频产业链关键组成部分,最后深入探讨超高清视频技术的应用领域,包括在电影、电视、文教娱乐等领域的应用,为超高清视频技术发展及应用场景拓展提供参考。展开更多
世界数字广播(Digital Radio Mondiale,DRM)是数字调幅广播的国际标准,其信息媒介为音频信号。介绍DRM系统中广泛采用的MPEG-4高级音频编解码(Advanced Audio Coding,AAC)标准及相关算法,包括心理声学模型、滤波、量化、无噪声编码及时...世界数字广播(Digital Radio Mondiale,DRM)是数字调幅广播的国际标准,其信息媒介为音频信号。介绍DRM系统中广泛采用的MPEG-4高级音频编解码(Advanced Audio Coding,AAC)标准及相关算法,包括心理声学模型、滤波、量化、无噪声编码及时域噪声整形的实现原理,分析音频解码的基本流程。在工程应用阶段,为了优化硬件接收器的设计方法,利用C++语言编写接收器软件,用于模拟硬件设备的功能。经测试,该软件工具能够有效解码DRM音频信息。展开更多
The different formats of codec stream carried in the radio access network and the core network make the double speech encoding/decoding necessary, which degrades the speech quality. Accordingly, codec negotiation tech...The different formats of codec stream carried in the radio access network and the core network make the double speech encoding/decoding necessary, which degrades the speech quality. Accordingly, codec negotiation technologies are necessary for unifying encoding/ decoding in the whole process. Transcoder Free Operation (TrFO), Tandem Free Operation (TFO), and network quality deciding technology are the leading codec negotiation technologies. The TrFO is a mechanism for optimum selection during the establishment of a call. It tries to establish connection between User Equipment (UE) without Transcoder (TC). Its successful fulfillment enables the efficient utilization of bandwidth. The TFO, a standby technology of TrFO, is the negotiation technology of an in-band codec. With it, the user codec stream is free from the compression and decompression by the voice codec, and the quality of voice can accordingly be improved. The network-quantity deciding technology adopts G.711 or G.729 flexibly according to the number of accessed calls. This allows the access of new calls while won’t increase the load of network too much.展开更多
Aiming at improving rate flexibility of the enhanced voice services (EVS) channel-aware mode for various VoIP applications, two new bit-rate channel-aware modes are proposed in this paper in addition to the existing 1...Aiming at improving rate flexibility of the enhanced voice services (EVS) channel-aware mode for various VoIP applications, two new bit-rate channel-aware modes are proposed in this paper in addition to the existing 13.2 kbit/s mode. Channel-aware mode uses forward error correction by transmitting re-encoded information redundantly for use when the original information is lost or discarded due to late arrival to the receiver. The primary frame bit rate is reduced for the redundant accommodation. A modified quantization scheme is proposed for core encoding regarding the quality degradation. Partial redundant coding is a simplification of that in the existing 13.2 kbit/s channel-aware mode due to the bit constraint. The objective evaluation results of PESQ show that the additional channel-aware modes achieve similar performance in improving the error robustness against missing packets as that of the existing 13.2 kbit/s mode. Multiple bit-rate modes can be dynamically selected in the communication system for more voice services in different bandwidths. On the other hand, optimal allocation based on real-time feedback can adapt to the rapidly-changing network environment as well as possible.展开更多
ITU-T G. 729 is the primarily recommended speech codec by H. 323 standard. This paper describes how to implement G. 729 codec in IP telephony gateway, and goes deep into the programming skills on TMS320C6201 DSP and o...ITU-T G. 729 is the primarily recommended speech codec by H. 323 standard. This paper describes how to implement G. 729 codec in IP telephony gateway, and goes deep into the programming skills on TMS320C6201 DSP and optimizing methods of program code to reduce the speech processing delay time of G. 729 codec. Due to adopting these optimizing methods and programming skills, we have implemented a high-speed speech codec that can process concurrently 20 voice channels with single TMS320C6201 chip in IP telephony gateway. Finally, the paper analyzes the performance results of ITU-T G. 729 codec based on TMS320C6201.展开更多
In this paper, we present a method using video codec technology to compress ECG signals. This method exploits both intra-beat and inter-beat correlations of the ECG signals to achieve high compression ratios (CR) and ...In this paper, we present a method using video codec technology to compress ECG signals. This method exploits both intra-beat and inter-beat correlations of the ECG signals to achieve high compression ratios (CR) and a low percent root mean square difference (PRD). Since ECG signals have both intra-beat and inter-beat redundancies like video signals, which have both intra-frame and inter-frame correlation, video codec technology can be used for ECG compression. In order to do this, some pre-process will be needed. The ECG signals should firstly be segmented and normalized to a sequence of beat cycles with the same length, and then these beat cycles can be treated as picture frames and compressed with video codec technology. We have used records from MIT-BIH arrhythmia database to evaluate our algorithm. Results show that, besides compression efficiently, this algorithm has the advantages of resolution adjustable, random access and flexibility for irregular period and QRS false detection.展开更多
文摘世界数字广播(Digital Radio Mondiale,DRM)是数字调幅广播的国际标准,其信息媒介为音频信号。介绍DRM系统中广泛采用的MPEG-4高级音频编解码(Advanced Audio Coding,AAC)标准及相关算法,包括心理声学模型、滤波、量化、无噪声编码及时域噪声整形的实现原理,分析音频解码的基本流程。在工程应用阶段,为了优化硬件接收器的设计方法,利用C++语言编写接收器软件,用于模拟硬件设备的功能。经测试,该软件工具能够有效解码DRM音频信息。
文摘The different formats of codec stream carried in the radio access network and the core network make the double speech encoding/decoding necessary, which degrades the speech quality. Accordingly, codec negotiation technologies are necessary for unifying encoding/ decoding in the whole process. Transcoder Free Operation (TrFO), Tandem Free Operation (TFO), and network quality deciding technology are the leading codec negotiation technologies. The TrFO is a mechanism for optimum selection during the establishment of a call. It tries to establish connection between User Equipment (UE) without Transcoder (TC). Its successful fulfillment enables the efficient utilization of bandwidth. The TFO, a standby technology of TrFO, is the negotiation technology of an in-band codec. With it, the user codec stream is free from the compression and decompression by the voice codec, and the quality of voice can accordingly be improved. The network-quantity deciding technology adopts G.711 or G.729 flexibly according to the number of accessed calls. This allows the access of new calls while won’t increase the load of network too much.
基金Supported by the International Cooperation Research Project between Ericsson(Sweden) and BIT
文摘Aiming at improving rate flexibility of the enhanced voice services (EVS) channel-aware mode for various VoIP applications, two new bit-rate channel-aware modes are proposed in this paper in addition to the existing 13.2 kbit/s mode. Channel-aware mode uses forward error correction by transmitting re-encoded information redundantly for use when the original information is lost or discarded due to late arrival to the receiver. The primary frame bit rate is reduced for the redundant accommodation. A modified quantization scheme is proposed for core encoding regarding the quality degradation. Partial redundant coding is a simplification of that in the existing 13.2 kbit/s channel-aware mode due to the bit constraint. The objective evaluation results of PESQ show that the additional channel-aware modes achieve similar performance in improving the error robustness against missing packets as that of the existing 13.2 kbit/s mode. Multiple bit-rate modes can be dynamically selected in the communication system for more voice services in different bandwidths. On the other hand, optimal allocation based on real-time feedback can adapt to the rapidly-changing network environment as well as possible.
基金Supported by the National Natural Science Foundation of China under grant!69773046
文摘ITU-T G. 729 is the primarily recommended speech codec by H. 323 standard. This paper describes how to implement G. 729 codec in IP telephony gateway, and goes deep into the programming skills on TMS320C6201 DSP and optimizing methods of program code to reduce the speech processing delay time of G. 729 codec. Due to adopting these optimizing methods and programming skills, we have implemented a high-speed speech codec that can process concurrently 20 voice channels with single TMS320C6201 chip in IP telephony gateway. Finally, the paper analyzes the performance results of ITU-T G. 729 codec based on TMS320C6201.
文摘In this paper, we present a method using video codec technology to compress ECG signals. This method exploits both intra-beat and inter-beat correlations of the ECG signals to achieve high compression ratios (CR) and a low percent root mean square difference (PRD). Since ECG signals have both intra-beat and inter-beat redundancies like video signals, which have both intra-frame and inter-frame correlation, video codec technology can be used for ECG compression. In order to do this, some pre-process will be needed. The ECG signals should firstly be segmented and normalized to a sequence of beat cycles with the same length, and then these beat cycles can be treated as picture frames and compressed with video codec technology. We have used records from MIT-BIH arrhythmia database to evaluate our algorithm. Results show that, besides compression efficiently, this algorithm has the advantages of resolution adjustable, random access and flexibility for irregular period and QRS false detection.
