GSM Speech Coder Indirect Identification Algorithm

نویسندگان

  • Rajko Svecko
  • Bojan Kotnik
  • Amor Chowdhury
  • Zdenko Mezgec
چکیده

This paper presents GSM speech coder indirect identification algorithm based on sending novel identification pilot signals through the GSM speech channel. Each GSM subsystem disturbs identification pilot, while speech coder uniquely changes the tempo-spectral characteristics of the proposed pilot signal. Speech coder identification algorithm identifies speech coder with the usage of robust linear frequency cepstral coefficient (LFCC) feature extraction procedure and fast artificial neural networks. First step of speech coder identification algorithm is the exact position detection of the identification pilot signal using normalized cross correlation approach. Next stage is time-domain windowing of the input signal to convolve each frame of the input speech signal and window spectrum. Consecutive step is a short-time Fast Fourier Transformation to produce the magnitude spectrum of each windowed frame. Further, a noise reduction with spectral subtraction based on spectral smoothing is carried out. In last steps we perform the frequency filtering and Discrete Cosine Transformation to receive 24 uncorrelated cepstral coefficients per frame as a result. Speech coder identification is completed with fast artificial neural network classification using the input feature vector of 24 LFCC coefficients, giving a result of identified speech coder. For GSM speech coder indirect identification evaluation, the standardized GSM ETSI bit-exact implementations were used. Furthermore, a set of custom tools was build. These tools were used to simulate and control various conditions in the GSM system. Final results show that proposed algorithm identifies the GSM-EFR speech coder with the accuracy of 98.85%, the GSM-FR speech coder with 98.71%, and the GSM-HR coder with 98.61%. These scores were achieved at various types of surrounding noises and even at very low SNR conditions.

برای دانلود متن کامل این مقاله و بیش از 32 میلیون مقاله دیگر ابتدا ثبت نام کنید

ثبت نام

اگر عضو سایت هستید لطفا وارد حساب کاربری خود شوید

منابع مشابه

Speech coding gsm pdf

speech coding in gsm pdf 260 bits.found that the use of GSM coding degrades significantly the identification and. Proposes an in-depth look at the influence of GSM speech coding on text. The full-rate GSM speech codec 2 is a lossy speech coding-decoding.In cellular communication technology, quality of voice output at destination depends on the channel condition. Bad channel condition will produ...

متن کامل

Wideband speech coding algorithm with application of discrete wavelet transform to upper band

In this paper, we propose a new wideband speech coder which combines the European standard of a narrowband speech coder, i.e., GSM-EFR, and a transform coder using the discrete wavelet transform. Input speech is first split into two bands with equal bandwidth. A subband coder with wavelet transformed speech is designed for a upper band coder, and a GSM-EFR coder is adopted as a lower band coder...

متن کامل

Proposed Modifications in ETSI GSM Full Rate Speech Codec in line with bitrates of GSM EFR Speech Codec and its Objective Evaluation of Performance using MATLAB

Today, the primary constrain in wireless communication system is limited bandwidth and power. Wireless systems involved in transmission of speech envisage that efficient and effective methods be developed (bandwidth usage & power) to transmit and receive the same while maintaining quality-of-speech, especially at the receiving end. Speech coding is a technique, since the era of digitization (di...

متن کامل

LOW−COMPLEXITY AUTOMATIC SPEAKER RECOGNITION IN THE COMPRESSED GSM AMR DOMAIN (WedAmOR2)

This paper presents an experimental implementation of a low−complexity speaker recognition algorithm working in the compressed speech domain. The goal is to perform speaker modeling and identification without the need of decoding the speech bitstream to extract speaker dependent features, thus saving important system resources, for instance, in mobile battery powered DSP devices. Bitstream valu...

متن کامل

An adaptive multi-rate speech coder for digital cellular telephony

We have developed an adaptive multi-rate (AMR) speech coder designed to operate under the GSM digital cellular full rate (22.8 kb/s) and half rate (11.4 kb/s) channels and to maintain high quality in the presence of highly varying background noise and channel conditions. Within each total rate, several codec modes with different source/channel bit rate allocations are used. The speech coders in...

متن کامل

ذخیره در منابع من


  با ذخیره ی این منبع در منابع من، دسترسی به آن را برای استفاده های بعدی آسان تر کنید

برای دانلود متن کامل این مقاله و بیش از 32 میلیون مقاله دیگر ابتدا ثبت نام کنید

ثبت نام

اگر عضو سایت هستید لطفا وارد حساب کاربری خود شوید

عنوان ژورنال:
  • Informatica, Lith. Acad. Sci.

دوره 21  شماره 

صفحات  -

تاریخ انتشار 2010