This paper describes a speech coder which has an input signal frame interval of 20ms percent in duration, contains 160 voice samples (8,000 samples/s) and 48 bits. A key feature of the coder is a novel Line Spectral F...
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ISBN:
(纸本)9783037858462
This paper describes a speech coder which has an input signal frame interval of 20ms percent in duration, contains 160 voice samples (8,000 samples/s) and 48 bits. A key feature of the coder is a novel Line Spectral Frequencies (LSF) quantization scheme, requiring only 19 bits per frame. This new coder, through algorithmic improvements, enhanced quantization techniques and new harmonic synthesis method, produces better speech quality at 2.4 kb/s transnission rate than the new U.S. Federal Standard MELP coder.
Transmission of data in Voice over Internet Protocol (VoIP) must be made secure and robust such that data should not be easily attacked by intruders. Main objective of the proposed system is to hide the secret informa...
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ISBN:
(纸本)9781538605691
Transmission of data in Voice over Internet Protocol (VoIP) must be made secure and robust such that data should not be easily attacked by intruders. Main objective of the proposed system is to hide the secret information in the silence part of speech signal for secure communication. Voice Activity Detection (VAD) Algorithm of ITU-T G.729B speech coder is performed to detect silence part of speech signal which is followed by Steganography for embedding and extraction of secret information. In order to evaluate the performance parameters for data hiding capacity in speech signal and for speech quality, the parameters like Perceptual Evaluation of speech Quality (PESQ), Absolute error (ABS), Root Mean Square Error (RAISE), Mean Square Error (MSE), Mean Optimum Score (MOS) are explored. Robustness for proposed hiding scheme is performed by introducing compression attack and resampling of speech signal.
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...
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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.
The Internet has revolutionized the telecommunication systems by supporting new applications and services. Voice over Internet Protocol (VoIP) is one of the most prominent telecommunication services based on the Inter...
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The Internet has revolutionized the telecommunication systems by supporting new applications and services. Voice over Internet Protocol (VoIP) is one of the most prominent telecommunication services based on the Internet Protocol (IP). The signal quality of the VoIP system depends on several factors such as networking conditions, coding processes, speech content and error correction schemes. The work in the present paper reviewed these issues, used for providing toll-quality communication service to the users over VoIP system. From the very beginning of transferring the voice data over packet switched networks, the journey of the packet based communications to modern VoIP and advancements to improve the service of the VoIP system has been summarized in this work. (C) 2013 Elsevier Ltd. All rights reserved.
Nowadays the number of mobile subscribers is increasing all over the world, so the system for the communication has to be improved. Mixed Excited Linear Prediction (MELP) algorithm is developed for reducing the bandwi...
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ISBN:
(纸本)9781479949816
Nowadays the number of mobile subscribers is increasing all over the world, so the system for the communication has to be improved. Mixed Excited Linear Prediction (MELP) algorithm is developed for reducing the bandwidth of the signal as well as transmit more data on a single channel. This results in increase in channel capacity. MELP is basically a speech coding method, relying on a speech Encoder and speech Decoder. The MELP speech coder reduces the redundancy of the signal and compresses it, which is represented by the MELP code. speech Decoder includes a Linear Predictive Coding (LPC) filter providing a synthesized speech at its output side in response to voice and unvoiced. MELP also reduces jitter voice. The bit rate of MELP is reducing the reserves of the code book and calculation complexity. This paper describes the bit rates of MELP coder can be reduced to as low as 2.4kbps without apparent damage to the synthetic speech quality.
Voice Conversion(VC) consists in modifying a source voice to a target speaker voice. In our approach, we modified only the Code excited linear Predictive(CELP) coder by introducing a pre-processing before the coder fo...
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Voice Conversion(VC) consists in modifying a source voice to a target speaker voice. In our approach, we modified only the Code excited linear Predictive(CELP) coder by introducing a pre-processing before the coder for the voice conversion. The decoder part of CELP was not modified. This allows maintaining the transmission rate. Our approach for conversion consists in separating the voiced and unvoiced frames, and thus two different conversion functions are associated. The Spectral Frequency Parameters LSF parameters are adopted to represent the vocal tract and Gaussian Mixture Models(GMM) are used to calculate the conversion functions. The pitch for the voiced frames is transformed by linear conversion. The model was tested for conversions between male and female voices.
Perceptual linear prediction (PLP) is widely used in speech recognition systems as a feature extraction method. Also code-Exited Linear Prediction coder (CELP) is one of the well known speech coders which widely used ...
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ISBN:
(纸本)9780769539256
Perceptual linear prediction (PLP) is widely used in speech recognition systems as a feature extraction method. Also code-Exited Linear Prediction coder (CELP) is one of the well known speech coders which widely used in communication systems. In this paper the application of PLP in speech coding has been discussed. In the first stage the parameters of formant synthesis filter are determined by applying PLP algorithms. Then these parameters are used in coder, code-exited linear prediction coder, to improve the efficiency of this kind of coder. The experiments show promising result in some cases.
3(rd) Generation Project Plan (3GPP) has standardized the Wideband Adaptive Multi-rate (AMR) speech coder for 3GPP-based International Mobile Telecommunications 2000 (IMT-2000) system. This paper proposed a new method...
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ISBN:
(纸本)0889865078
3(rd) Generation Project Plan (3GPP) has standardized the Wideband Adaptive Multi-rate (AMR) speech coder for 3GPP-based International Mobile Telecommunications 2000 (IMT-2000) system. This paper proposed a new method to reduce the complexity of wideband AMR speech coder. Using this method the complexity can be reduced to half of original method with no speech quality degradation. We describe the efficient software scheme and sophisticated hardware design for real-time implementation of wideband AMR speech. We verified complexity reduced algorithm performance using SNR and ITU-T P.862 (PESQ) measure. We also evaluated its real-time operating performance using hard ware test system.
In this paper, the latest wideband vocoder standard adopted by the cdma2000 standardization body, 3GPP2, is described. Christened Enhanced Variable Rate Codec- Wideband (EVRC-WB), the proposed codec encodes wideband s...
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ISBN:
(纸本)1424407281
In this paper, the latest wideband vocoder standard adopted by the cdma2000 standardization body, 3GPP2, is described. Christened Enhanced Variable Rate Codec- Wideband (EVRC-WB), the proposed codec encodes wideband speech (16 KHz sampling frequency) at a maximum bit-rate of 8.55 kbit/s. EVRC-WB is based on a split band coding paradigm in which two different coding models are used for the signal components in the low frequency (LF) (0-4 KHz) and the high frequency (HF) (3.5-7 KHz) bands. The coding model used for the former is based on the EVRC-B narrowband (0-4 KHz) codec, modified to encode the LF band signal at a maximum bitrate of 7.75 kbit/s. The HF band coding model is a LPC based coding scheme where the excitation is derived from the coded LF band excitation using non-linear processing. Mean opinion scores from 3GPP2 characterization tests are provided to demonstrate that the EVRC-WB codee (8.55 kbit/s, max.) performs statistically significantly better than the Adaptive Multirate Wideband (12.65 kbit/s, max.).
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