摘要：
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.
摘要：
This paper puts forward an effective way to suppress acoustic feedback based on TI's TMS320C6713 and using frequency shift.It adopts the Codec chip TLV320AIC23 to capture and play back speech signal, which can connect with DSP seamlessly. Then completes real-time implementation on DSP based on simulating and suppressing screaming in Matlab ideally. The result is also evaluated subjectively and objectively. It turns out that it can suppress the interference of the regenerative reverberation, enhance the amplification gain greatly and improve frequency response characteristics and sound articulation obviously.%提出了一種基于TI公司TMS320C6713 DSP和移頻法抑制聲反饋的有效方法.該方法采用能與之無縫連接的TLV320AIC23 Codec芯片作為語音采集和回放工具,然后基于在Matlab進(jìn)行仿真達(dá)到抑制嘯叫相當(dāng)理想的基礎(chǔ)上完成了在DSP上的實時實現(xiàn).最后,采用主觀法和客觀法評估了輸出語音的質(zhì)量.結(jié)果表明,該方法能有效抑制再生混響干擾,明顯提高了擴(kuò)聲增益,且顯著改善了頻響特性和聲音清晰度.
摘要：
Nowadays, from home monitoring to large airport security, a lot of digital video surveillance systems have been used. Digital surveillance system usually requires streaming video processing abilities. As an advanced video coding method, H.264 is introduced to reduce the large video data dramatically (usually by 70X or more). However, computational overhead occurs when coding and decoding H.264 video. In this paper, a System-on-a-Chip (SoC) based hardware acceleration solution for video codec is proposed, which can also be used for other software applications. The characteristics of the video codec are analyzed by using the profiling tool. The Hadamard function, which is the bottleneck of H.264, is identified not only by execution time but also another two attributes, such as cycle per loop and loop round. The Co-processor approach is applied to accelerate the Hadamard function by transforming it to hardware. Performance improvement, resource costs and energy consumption are compared and analyzed. Experimental results indicate that 76.5% energy deduction and 8.09X speedup can be reached after balancing these three key factors.
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