• Title/Summary/Keyword: speech codec

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Real-Time Implementation of AMR Speech Codec Using TMS320VC5510 DSP (TMS320VC5510 DSP를 이용한 AMR 음성부호화기의 실시간 구현)

  • Kim, Jun;Bae, Keun-Sung
    • MALSORI
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    • no.65
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    • pp.143-152
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    • 2008
  • This paper focuses on the real time implementation of an adaptive multi-rate (AMR) speech codec, that is a standard speech codec of IMT-2000, using the TMS320VC5510. The series of TMS320VC55x is a 16-bit fixed-point digital signal processor (DSP) having low power consumption for the use of mobile communications by Texas Instruments (TI) corporation. After we analyze the AMR algorithm and source code as well as the structure and I/O of 7MS320VC55x, we carry out optimizing the programs for real time implementation. The implemented AMR speech codec uses 55.2 kbyte for the program memory and 98.3 kbyte for the data memory, and it requires 709,878 clocks, i.e. about 3.5 ms, for processing a frame of 20 ms speech signal.

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A Study on the Development of the Real-Time G.723.1 Speech Codec Using a Fixed-Point DSP(ADSP-2181) (고정소수점 DSP(ADSP-2181)을 이용한 실시간 G.723.1 음성부호화기 개발에 관한 연구)

  • Park, Jung-Jae;Chung, Ik-Joo
    • Speech Sciences
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    • v.3
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    • pp.177-186
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    • 1998
  • This paper describes the procedure of implementing a real-time speech codec, G.723.1 which was developed by DSP Group and standardized by ITU-T, using fixed-point DSP, ADSP-2181. This codec has two bit rates associated with it, 5.3 and 6.3 kbit/s. We implemented only one bit rate, 6.3 kbit/s, of the two with fixed-point 32-bit precision. According to the result of the experiment, the amount of computational burden is about 55 MIPS and its quality is similar to the result of the PC simulation with floating-point arithmetic. In this paper, we proposed a method to use a fixed-point DSP and a procedure for developing a real-time speech codec using DSPs and finally developed a G.723.l speech codec for ADSP-2181.

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Bandwidth Scalable Wideband Speech Codec (대역폭 계층 구조의 광대역 음성 부호차기 개발)

  • 이우석;손창용;이영범;박호종
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.6
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    • pp.481-487
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    • 2004
  • In this paper. the structure of bandwidth scalable wideband speech codec and its high-band codec are proposed. In the high-band codec. the signal is divided into frequency bands. and each band is quantized in DCT domain. The DCT coefficients are splitted into magnitude and sign, and each is quantized independently by a specialized method based on its characteristics. In addition. the quantized gain parameter in the low-band codec is utilized in the high-band codec for an enhanced performance. The bandwidth scalable wideband speech codec using G.729E for low-band and the proposed codec for high-band is developed, and it is confirmed that the proposed codec has better subjective performance than 24kbps G.722.1.

High-Band Codec for Bandwidth Scalable Wideband Speech Codec (대역폭 계층 구조의 광대역 음성 부호화기를 위한 상위 대역 부호화기 연구)

  • Kim Youngvo;Jeong Byounghak;Son Chang-Yong;Sung Ho-Sang;Park Hochong
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.7
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    • pp.395-401
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    • 2005
  • In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.

Experiment of VoIP Transmission with AMR Speech Codec in Wireless LAN (무선랜 환경에서 AMR 음성부호화기를 적용한 VoIP 전송 실험)

  • Shin, Hye-Jung;Bae, Keun-Sung
    • Speech Sciences
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    • v.11 no.4
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    • pp.67-73
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    • 2004
  • Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

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An Embedded ACELP Speech Coding Based on the AMR-WB Codec

  • Byun, Kyung-Jin;Eo, Ik-Soo;Jeong, Hee-Bum;Hahn, Min-Soo
    • ETRI Journal
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    • v.27 no.2
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    • pp.231-234
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    • 2005
  • This letter proposes a new embedded speech coding structure based on the Adaptive Multi-Rate Wideband (AMR-WB) standard codec. The proposed coding scheme consists of three different bitrates where the two lower bitrates are embedded into the highest one. The embedded bitstream was achieved by modifying the algebraic codebook search procedure adopted for the AMR-WB codec. The proposed method provides the advantage of scalability due to the embedded bitstream, while it inevitably requires some additional computational complexity for obtaining two different code vectors of the higher bitrate modes. Compared to the AMR-WB codec, the embedded coder shows improved speech qualities for two higher bitrate modes with a slightly increased bitrate caused by the decreased coding efficiency of the algebraic codebook.

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16kbps Windeband Sideband Speech Codec (16kbps 광대역 음성 압축기 개발)

  • 박호종;송재종
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.1
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    • pp.5-10
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    • 2002
  • This paper proposes new 16 kbps wideband speech codec with bandwidth of 7 kHz. The proposed codec decomposes the input speech signal into low-band and high-band signals using QMF (Quadrature Mirror Filter), then AMR (Adaptive Multi Rate) speech codec processes the low-band signal and new transform-domain codec based on G.722.1 wideband cosec compresses the high-band signal. The proposed codec allocates different number of bits to each band in an adaptive way according to the property of input signal, which provides better performance than the codec with the fixed bit allocation scheme. In addition, the proposed cosec processes high-band signal using wavelet transform for better performance. The performance of proposed codec is measured in a subjective method. and the simulations with various speech data show that the proposed coders has better performance than G.722 48 kbps SB-ADPCM.

Performance Comparison on Speech Codecs for Digital Watermarking Applications

  • Mamongkol, Y.;Amornraksa, T.
    • Proceedings of the IEEK Conference
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    • 2002.07a
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    • pp.466-469
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    • 2002
  • Using intelligent information contained within the speech to identify the specific hidden data in the watermarked multimedia data is considered to be an efficient method to achieve the speech digital watermarking. This paper presents the performance comparison between various types of speech codec in order to determine an appropriate one to be used in digital watermarking applications. In the experiments, the speech signal encoded by four different types of speech codec, namely CELP, GSM, SBC and G.723.1codecs is embedded into a grayscale image, and theirs performance in term of speech recognition are compared. The method for embedding the speech signal into the host data is borrowed from a watermarking method based on the zerotrees of wavelet packet coefficients. To evaluate efficiency of the speech codec used in watermarking applications, the speech signal after being extracted from the attacked watermarked image will be played back to the listeners, and then be justified whether its content is intelligible or not.

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Delayless MDCT for Scalable Speech Codec (계층구조 음성 부호화기를 위한 지연 없는 MDCT 구조)

  • Sung, Ho-Sang;Park, Ho-Chong
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.3
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    • pp.102-108
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    • 2007
  • A high-Performance scalable speech codec generally requires a very low-rate first layer and a fine granule second layer, and this codec can be implemented with the harmonic codec and the MDCT-based transform codec for each layer. In this structure, however. each codec requires independent frequency transform and the time delay of each codec is accumulated. resulting in long time delay for the overall codec. In this paper, new MDCT structure in the second layer is Proposed. where MDCT is forced to share the look-ahead region of the first layer in order to prevent the time delay accumulation and the resulting functional error of MDCT is analyzed and removed after IMDCT The Proposed delayless MDCT requires no additional bits and Provides the equivalent coding performance with the reduced time delay, yielding a meaningful enhancement of the overall codec.

A Practical Implementation of the LTJ Adaptive Filter and Its Application to the Adaptive Echo Canceller (LTJ 적응필터의 실용적 구현과 적응반향제거기에 대한 적용)

  • Yoo, Jae-Ha
    • Speech Sciences
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    • v.11 no.2
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    • pp.227-235
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    • 2004
  • In this paper, we proposed a new practical implementation method of the lattice transversal joint (LTJ) adaptive filter using speech codec's information. And it was applied to the adaptive echo cancellation problem to verify the efficiency of the proposed method. Realtime implementation of the LTJ adaptive filter is very difficult due to high computational complexity for the filter coefficients compensation. However, in case of using speech codec, complexity can be reduced since linear predictive coding (LPC) coefficients are updated each frame or sub-frame instead of every sample. Furthermore, LPC coefficients can be acquired from speech decoder and transformed to the reflection coefficients. Therefore, the computational complexity for updates of the reflection coefficients can be reduced. The effectiveness of the proposed LTJ adaptive filter was verified by the experiments about convergence and tracking performance of the adaptive echo canceller.

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