• Title/Summary/Keyword: CODEC

Search Result 694, Processing Time 0.034 seconds

Analysis and Implementation of Speech/Music Classification for 3GPP2 SMV Codec Employing SVM Based on Discriminative Weight Training (SMV코덱의 음성/음악 분류 성능 향상을 위한 최적화된 가중치를 적용한 입력벡터 기반의 SVM 구현)

  • Kim, Sang-Kyun;Chang, Joon-Hyuk;Cho, Ki-Ho;Kim, Nam-Soo
    • The Journal of the Acoustical Society of Korea
    • /
    • v.28 no.5
    • /
    • pp.471-476
    • /
    • 2009
  • In this paper, we apply a discriminative weight training to a support vector machine (SVM) based speech/music classification for the selectable mode vocoder (SMV) of 3GPP2. In our approach, the speech/music decision rule is expressed as the SVM discriminant function by incorporating optimally weighted features of the SMV based on a minimum classification error (MCE) method which is different from the previous work in that different weights are assigned to each the feature of SMV. The performance of the proposed approach is evaluated under various conditions and yields better results compared with the conventional scheme in the SVM.

Improving Speech Quality of VoIP by Packet Prioritization (패킷 중요도 결정에 의한 VoIP 통화 품질 향상 기술)

  • Yoon, Jae-Yul;Park, Ho-Chong
    • The Journal of the Acoustical Society of Korea
    • /
    • v.29 no.5
    • /
    • pp.347-353
    • /
    • 2010
  • In VoIP system, the speech quality is seriously degraded due to packet loss, and the degree of degradation by each packet loss depends on the characteristics of the corresponding packet. Therefore, it is possible to improve the speech quality of VoIP by selectively controlling the packet to be lost during transmission based on the expected degradation by the loss of each packet. In this paper, a new scheme to improve speech quality of DiffServ-based VoIP by assigning priority to each packet is proposed, and a method to determine the priority of each packet is developed. The performance of proposed method was measured in packet loss environment based on Gilbert model, and it was verified both objectively and subjectively that the speech quality is improved by the proposed method.

Real-time Implementation of the AMR-WB+ Audio Coder using ARM Core(R) (ARM Core(R)를 이용한 AMR-WB+ 오디오 부호화기의 실시간 구현)

  • Won, Yang-Hee;Lee, Hyung-Il;Kang, Sang-Won
    • Journal of the Institute of Electronics Engineers of Korea SP
    • /
    • v.46 no.3
    • /
    • pp.119-124
    • /
    • 2009
  • In this paper, AMR-WB+ audio coder is implemented, in real-time, using Intel 400MHz Xscale PXA250 with 32bit RISC processor ARM9E-J(R)core. The assembly code for ARM9E-J(R)core is developed through the serial process of C code optimization, cross compile, assembly code manual optimization and adjusting the optimized code to Embedded Visual C++ platform. C code is trimmed on Visual C++ platform. Cross compile and assembly code manual optimization are performed on CodeWarrior with ARM compiler. Through these stages the code for both ARM EVM board and PDA is implemented. The average complexities of the code are 160.75MHz on encoder and 33.05MHz on decoder. In case of static link library(SLL), the required memories are 65.21Kbyte, 32.01Kbyte and 279.81Kbyte on encoder, decoder and common sources, respectively. The implemented coder is evaluated using 16 test vectors given by 3GPP to verify the bit-exactness of the coder.

Channel Condition Adaptive Error Concealment using Scalability Coding (채널상태에 적응적인 계층 부호화를 이용한 오류 은닉 방법 연구)

  • Han Seung-Gyun;Park Seung-Ho;Suh Doug-Young
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.29 no.1B
    • /
    • pp.8-17
    • /
    • 2004
  • In this paper: we propose the adaptive error concealment technique for scalable video coding over wireless network error prove environment. We prove it is very effective that Error concealment techniques proposed in this paper are applied to scalable video data. In this paper, we propose two methods of error concealment. First one is that the en·or is concealed using the motion vector of base layer and previous VOP data. Second one is that according to existence of motion vector in error position, the error is concealed using the same position data of base layer when the motion vector is existing otherwise using the same position data of previous VOP when the motion vector is 0(zero) adaptively. We show that according to various error pattern caused by condition of wireless network and characteristics of sequence, we refer decoder to base layer data or previous enhancement layer data to effective error concealment. Using scalable coding of MPEG-4 In this paper, this error concealment techniques are available to be used every codec based on DCT.

Error Concealment of MPEG-2 Intra Frames by Spatiotemporal Information of Inter Frames (인터 프레임의 시공간적 정보를 이용한 MPEG-2 인트라 프레임의 오류 은닉)

  • Kang, Min-Jung;Ryu, Chul
    • Journal of the Institute of Convergence Signal Processing
    • /
    • v.4 no.2
    • /
    • pp.31-39
    • /
    • 2003
  • The MPEG-2 source coding algorithm is very sensitive to transmission errors due to using of variable-length coding. When the compressed data are transmitted, transmission errors are generated and error correction scheme is not able to be corrected well them. In the decoder error concealment (EC) techniques must be used to conceal errors and it is able to minimize degradation of video quality. The proposed algorithm is method to conceal successive macroblock errors of I-frame and utilize temporal information of B-frame and spatial information of P-frame In the previous GOP which is temporally the nearest location to I-frame. This method can improve motion distortion and blurring by temporal and spatial errors which cause at existing error concealment techniques. In network where the violent transmission errors occur, we can conceal more efficiently severe slice errors. This algorithm is Peformed in MPEG-2 video codec and Prove that we can conceal efficiently slice errors of I-frame compared with other approaches by simulations.

  • PDF

An Image Concealment Algorithm Using Fuzzy Inference (퍼지 추론을 이용한 영상은닉 알고리즘)

  • Kim, Ha-Sik;Kim, Yoon-Ho
    • Journal of Advanced Navigation Technology
    • /
    • v.11 no.4
    • /
    • pp.485-492
    • /
    • 2007
  • In this paper, we propose the receiver block error detection of the video codec and the image concealment algorithm using fuzzy inference. The proposed error detection and concealment algorithm gets SSD(Summation of Squared Difference) and BMC(Boundary Matching Coefficient) using the temporal and spatial similarity between corresponded blocks in the two successive frames. Proportional constant, ${\alpha}$, for threshold value, TH1 and TH2, is decided after fuzzy data is generated by each parameter. To examine the propriety of the proposed algorithm, random errors are inserted into the QCIF Susie standard image, then the error detection and concealment performance is simulated. To evaluate the efficiency of the algorithm, image quality is evaluated by PSNR for the error detection and concealed image by the existing VLC table and by the proposed method. In the experimental results, the error detection algorithm could detect all of the inserted error, the image quality is improved over 15dB after the error concealment compare to existing error detection algorithm.

  • PDF

Non-Reference P Frame Coding for Low-Delay Encoding in Internet Video Coding (IVC의 저지연 부호화 모드를 위한 비참조 P 프레임의 부호화 기법)

  • Kim, Dong-Hyun;Kim, Jin-Soo;Kim, Jae-Gon
    • Journal of Broadcast Engineering
    • /
    • v.19 no.2
    • /
    • pp.250-256
    • /
    • 2014
  • Non-reference P frame coding is used to enhance coding efficiency in low-delay encoding configuration of Internet Video Coding (IVC), which is being standardized as a royalty-free video codec in MPEG. The existing method of non-reference P frame coding which was adopted in the reference Test Model of IVC (ITM) 4.0 adaptively applies a non-reference P frame with a fixed coding structure based on the magnitude of motion vectors (MVs), however, which unexpectedly degrades the coding efficiency for some sequences. In this paper, the existing non-reference P frame coding is improved by changing non-reference P frame coding structure and applying a new adaptive method using the ratio of the amount of generated bits of non-reference frames to that of reference frames as well as MVs. Experimental results show that the proposed non-reference P frame coding gives 6.6% BD-rate bit saving in average over ITM 7.0.

Implementation of Implantable Bluetooth Bio-telemetry System for Transmitting Acoustic Signals in the Body with Wireless Recharging Function (무선 충전 가능한 블루투스 방식의 체내 음향신호 전송용 이식형 바이오 텔레메트리 시스템 구현)

  • Lee, Sang-June;Kim, Myoung Nam;Lee, Jyung Hyun;Lim, Hyung-Gyu;Cho, Jin-Ho
    • Journal of Korea Multimedia Society
    • /
    • v.18 no.5
    • /
    • pp.652-662
    • /
    • 2015
  • It is necessary to develop small, implantable bio-telemetry systems which can measure and transmit patients' bio-signals from internal body to external receiver. When measuring bio-signals, like electrical bio-signals, acoustic bio-signal measurement has also a big clinical usefulness. But, sound signal has larger frequency bandwidth than any other bio-signals. When considering these issues, a wireless telemetry system which has rapid data transmission rate proportional to wide frequency bandwidth is necessary to be developed. The bluetooth module is used to overcome the data rate limitation caused by the large frequency bandwidth. In this paper, a novel multimedia bluetooth biotelemetry system was developed which consists of transmitter module located in the body and receiver device located outside of the body. The transmitter consists of microphone, bluetooth, and wireless charging device. And the receiver consists of bluetooth and codec system. The sound inside the skin is captured by microphone and sent to receiver by bluetooth while charging. The wireless charging system constantly supplies the electric power to the system. To verify the performance of the developed system, an in vitro experiment has been performed. The results show that the proposed biotelemetry system has ability to acquire the sound signals under the skin.

Speech Packet Transmission Using the AMR-WB Coder with FEC (FEC기능을 추가한 AMR-WB 음성 부호화기를 이용한 음성 패킷 전송)

  • 황정준;이인성
    • Journal of the Institute of Electronics Engineers of Korea TC
    • /
    • v.40 no.11
    • /
    • pp.63-71
    • /
    • 2003
  • This paper suggests the packet loss recovery method to communicate in real time in the Internet. To reduce the effects of packet loss, Forward Error Correction (FEC) that adds redundant information to voice packets can be used. Adaptive Multi Rate Wideband(AMR-WB) codec which is recently selected by the Third Generation Partnership Project(3GPP) for GSM and the third generation mobile communication WCDMA system and has also been standardized in ITU-T for providing wideband speech services is used. The major cause for speech qualitly degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme if evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.

QCELP Implementation on TMS320C30 DSP Board TMS320C30 DSP를 이용한 QCELP Codec의 실현

  • Han, Kyong-Ho
    • The Journal of the Acoustical Society of Korea
    • /
    • v.14 no.1E
    • /
    • pp.83-87
    • /
    • 1995
  • The implementation of the voice dodec is imjplemented by using TMS320C30, which is the floating point DSP chip from Texas Instrument. QCELP (Qualcomm Code Excited Linear Prediction) is used to encode and decode the voice. The QCELP code is implemented by the TMS320C30 C-dode. The DSP board is controlled by the PC. The PC program tranfors the voice file from and to the DSP board, which is also implemented by C-code. The voice is encoded by the DSP board and the encoded data is transferred to PC to be stored as a file. To hear the voice. the voice data file is sent to DSP board and decoded to synthesize audible voice. Two flags are used by both programs to notify the status of the operation. By checking the flags, DSP and PC decides when the voice data is transferred between them.

  • PDF