When the bit rate of a compressed video sequence is reduced by a frequency domain transcoder system, the rate control scheme plays a very important role in maintaining consistent video quality. In this paper, we propose an efficient rate control scheme based on the complexity of MB (Macro Block) while conventional transcoding schemes use that of a picture. Since the frequency domain transcoder has to calculate the spatial activity of MB to adjust the quantization step, a process of converting the DCT (Discrete Cosine Transform) data into spatial one is required. The proposed scheme calculates the spatial activity from DCT data without converting them to pixel domain.

Video transcoding technique is an efficient mechanism to deliver visual contents to a variety of users who have different network conditions or terminal devices with different display capabilities. In this paper, we propose two types of transcoding methods for adapting the bitrate of streaming video to the bandwidth of the transmission channel; spatial resolution reduction (SRR) transcoding and temporal resolution reduction (TRR) transcoding. The two transcoding methods are alternatively operated according to the requirements of users. Experimental results show that the proposed transcoding methods can preserve image quality while transcoding to the low bitrate.

When the frame size is downscaled for video transcoding, the new motion vector (MV) must be computed. This paper presents an algorithm to utilize the activity measurement by DC value and the number of non-zero quantized DCT coefficients in the residual macroblock to compose the motion vector. It can reduce the complexity for motion estimation and improve the performance of the spatial domain video transcoder.

The latest video coding standard, H.264/AVC, adopts 44 approximate transform instead of 88 discrete cosine transform (DCT) to avoid the inverse transform mismatch problem. However, that is only one of the factors that make it difficult to transcode pre-coded video contents with the previous standards to H.264/AVC in the common domain without causing cascaded pixel-domain transcoding. In this paper, to support the existent DCT-domain transcoding schemes and to reduce computational complexity, we propose an efficient algorithm that converts the quantized 88 DCT block into four newly quantized 44 transformed blocks. The experimental results show that the proposed scheme reduces computational complexity by 5-11% and improves video quality by 0.1-0.5 dB compared with the cascaded pixel-domain transcoding scheme that exploits inverse quantization (IQ), inverse DCT (IDCT), DCT, and re-quantization (re-Q).

Recent microprocessors have included SIMD (single instruction multiple data) extensions into their instruction set architecture to improve the performance of multimedia applications. SIMD instructions speed up the execution of programs but pose lots of challenges to software developers. An efficient matrix-based splitter (or merger), which can split an N N 2-D DCT block into four N/2 N/2 or two N N/2 (or N/2 N) 2-D DCT blocks (or merger small size blocks into a large size one), specialized for SIMD architectures is presented in this paper. The programming-level complexity of the proposed methods is lower than that of the direct approach. Furthermore, even without using SIMD instructions, the algorithmic-level complexity of the proposed DCT splitter/merger is still lower than that of the direct one and is the same as that of the most efficient approach existed in the literature. When N = 8, our method can be applied to act as a transcoder between the latest video coding standards AVC/H.264 and the older ones, such as MPEG-1, MPEG-2 and MPEG-4 part 2. We also provide the image quality tests to show the performance of the proposed 2-D DCT splitter and merger.

In this paper, we propose a speech coding translation scheme by transferring coding parameters between GSM half rate and G.729 coders. Compared to the conventional decode-then-encode (DTE) scheme, the proposed parameter conversions provide speech interoperability between mobile and IP networks with reducing computational complexity and coding delay. Simulation results show that the proposed methods can reduce about 30% computational load and coding delay acquired in the target encoders and achieve almost imperceptible degradation in performance.

The current Internet does not offer any quality of service guarantees or support to Internet multimedia applications such as Internet telephony and video-conferencing, due to the best-effort nature of the Internet. Their performance may be adversely affected by network congestion. Also, since these applications commonly employ the UDP transport protocol, which lacks congestion control mechanisms, they may severely overload the network and starve other applications. We present an overview of recent research efforts in developing adaptive delivery models for Internet multimedia applications, which dynamically adjust the transmission rate according to network conditions. We classify the approaches used to develop adaptive delivery models with brief descriptions of representative research work. We then evaluate the approaches based on important design issues and performance criteria, such as the scalability of the control mechanism, responsiveness in detecting and reacting to congestion, and ability to accommodate receiver heterogeniety. Some conclusions are developed regarding the suitability of particular design choices under various conditions.