Ninad Bhatt

Artificial Bandwidth Extension of Speech & Its Applications in Wireless Communication Systems: A Review

In the recent scenario of wired and wireless communication systems, amongst many reasons for the overall degradation of recovered speech quality at receiver, one of the major reasons to be considered is utilization of Narrow Band (NB) end devices and NB transmission medium supporting bandwidth of 300 Hz-3400Hz. The inherent drawback of such NB speech signal is it sounds muffled and thin because of absence of High Band (HB) spectral components. With state of the art development of communication technologies and with increased availability of end terminals capable of transmitting and receiving Wide Band (WB) signals (having bandwidth from 50 Hz to 7000 Hz), end users prefer to listen to WB speech. In order to offer a fully WB communication over wired and wireless media, both end devices and network need to be made WB compatible. A long transition period has been elapsed for upgrading existing NB systems (both end terminals and network) to fully WB compatible systems. In-between new methods have been developed to artificially extend the bandwidth of NB telephonic speech at receiver for improving the quality of recovered speech. A major task of Artificial Bandwidth Extension (ABE) is to reconstruct missing WB spectral components at receiver with the use of available NB speech. This paper discusses motivation for developing ABE algorithm along with exhaustive comparative studies of implementing it with various approaches. Issues and limitations related to real time implementation of ABE algorithm are also addressed. Alternative approaches like usage of ABE with side information transmission along with coded NB speech are also demonstrated. Main objective of ABE with side information is to extract WB spectral components from WB input speech and to embed these derived spectral components into coded NB speech signal and finally transmit them onto a NB channel. Reverse procedure can be carried out at receiver to artificially produce WB speech. Here, it is to be noted that transmission channel is NB whereas end terminals are made WB compatible so this method provides alternative solution to coexisting state of the art WB coders (which require WB channels) while offering comparable speech quality and giving natural sounding in terms of intelligibility and naturalness.

Adaptive Multi-rate Wideband Speech Codec Based on CELP Algorithm: Architectural Study, Implementation & Performance Analysis

This paper investigates application of Code Excited Linear Prediction algorithm on Adaptive Multi Rate Wideband coder. The proposed coder can adaptively change its bit-rate based on C/I ratio depending on channel conditions. The coder has nine bit-rates from 6.6 kbps to 23.85 kbps. An e-test bench using MATLAB is created to implement proposed coder and series of simulations are carried out to judge the performance of implemented coder using Subjective and Objective analysis. Simulation results clearly advocate that it is possible to produce variable bit rate (by tuning to channel conditions) in CELP coder by affecting coefficients of coder while still maintaining comparable speech quality with reference to AMR WB coder standardized by 3GPP and ITU-T [5]. It is also evident from the simulation results that Signal to Noise Ratio (SNR), Segmented SNR, Perceptual Evaluation of Speech Quality (PESQ) and Mean Opinion Score (MOS) increases with increase in bit rates of proposed coder and Absolute Error (Abs Err), Mean Square Error (MSE), Root Mean Square Error (RMSE) reduces with increase in bitrates.

Implementation of variable bitrate data hiding techniques on standard and proposed GSM 06.10 full rate coder and its overall comparative evaluation of performance

Paper deals with implementation of variable bit rate steganographic data transmission over ETSI GSM 06.10 FR coder at five different bitrates. Then, few modifications are suggested in Regular Pulse Excitation section of ETSI GSM FR coder which ultimately claims to produce state of the art proposed GSM FR coder. In contrast with ETSI GSM FR coder, proposed coder also exhibits same bit rate steganographic data transmission. Here, in order to facilitate the same, few RPE pulses are identified and being utilized for embedding and hiding the information bits into them. Key element of this research is to allow for joint speech coding and data hiding and that is accomplished with two different approaches like Fixed and Joint Approach. These both approaches are implemented on both Standard and Proposed coders for their overall analytical evaluation of performance using Subjective (Mean opinion Score and Degraded MOS) and Objective (Perceptual Evaluation of Speech Quality) analysis. Small data information is represented as stego signal which can be embedded over different encoded wave files (chosen from NOIZEUS corpus) that serve as carrier signal. Simulation results for both coders reveal the trade off between data embedding rate and recovered speech quality (for both approaches). It is quite evident from both Subjective and Objective analysis that proposed coder offers comparable performance at the same time with lesser simulation delay because of its inherent constructional difference. It remains the fact that for both the coders, Joint approach performs better but at the cost of more simulation delay.

Proposed Modification in ETSI GSM 06.10 Full Rate Speech Codec for High Rate Data Hiding and its Objective Evaluation of Performance Using Simulink

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 (digital) and computerization (computational and processing horsepower - DSP) that has been a material-of-research for quite some time amongst the scientific and academic community. This paper proposes modifications in grid selection strategy in Regular Pulse Excitation section of ETSI GSM 06.10 Full Rate 13 kbps coder [6] so that there is an overall 1.8 kbps (36bits / each 20ms frame) reduction in bit-rate which can be utilized for high bit rate data hiding. These steganographic 36bits are appended in class 1B (bit d146-d181) as per 05.03[7] to finally produce bit stream of 260 bits for each frame. So, proposed modifications in GSM FR allow joint data hiding and speech coding. Watermark data like text, audio and image can be reliably transmitted at a rate of 1.8 kbps with a small effect on Objective speech quality (as can be witnessed from the obtained results with text file chosen in our analysis) and with reasonable computational complexity [1]. Here, both ETSI GSM 06.10 FR coder and proposed steganographic GSM FR coders are implemented using Simulink model in MATLAB and then Objective analysis comparison between both coders are carried out using set of tables and graphs. Apart from hiding data, the proposed modifications in GSM FR is useful for overall reduction in codec bit-rate and provides room for better error concealment at channel coding.

Simulation and overall comparative evaluation of performance between different techniques for high band feature extraction based on artificial bandwidth extension of speech over proposed global system for mobile full rate narrow band coder

This paper addresses a novel approach to investigate, study and simulate computation of high band (HB) feature extraction based on linear predictive coding (LPC) and mel frequency cepstral coefficient (MFCC) techniques. Further, HB features are embedded into encoded bitstream of proposed global system for mobile (GSM) full rate (FR) 06.10 coder using joint source coding and data hiding before being transmitted to receiving terminal. At receiver, HB features are extracted to reproduce HB portion of speech and for the same different extension of excitation techniques are applied and their results evaluated in terms of quality (intelligibility and naturalness) and bandwidth. MATLAB based e-test bench is created for implementing the proposed artificial bandwidth extension (ABE) coder following series of simulations, that are carried out to discover and gain insight about the performance of it using subjective [mean opinion score (MOS)] and objective [perceptual evaluation of speech quality (PESQ)] analysis. The results obtained for both the analyses advocate that proposed ABE coder outperforms proposed GSM FR NB (legacy GSM FR) coder. While the fact remains that, compared to LPC based parameterizations over ABE coder, MFCC parameterization results in higher speech intelligibility which is evident from obtained slightly better PESQ and MOS scores.

Implementation and Overall Performance Evaluation of CELP Based GSM AMR NB Coder over ABE

In todays wireless communication system, quality of decoded speech at receiving end is found muffled and thin, mainly attributed to inherent band limitation (300-3400 Hz) and power constraints. In order to obtain toll quality of recovered speech in terms of intelligibility and naturalness in wireless systems, Narrowband (NB) speech coders should be upgraded to its counterpart Wideband (WB) coders (50-7000Hz). In the meantime, a novel and backward compatible solution is proposed that claims to artificially extend bandwidth of NB speech to WB at receiving end, popular as Artificial Bandwidth Extension (ABE). Out of many techniques which aim to mitigate the effect of the ever unpredictable channel conditions, Adaptive Multi Rate (AMR) NB coder is considered to be one of the potential candidates. Selection of particular bit rate mode (out of all eight bit rate modes between 4.75 kbps and 12.2 kbps) solely depends upon the channel condition. This paper discusses development of ABE algorithm for Code Excited Linear Prediction (CELP) based GSM AMR NB coder, and for the same, MATLAB based e-test bench is created for simulation. Such series of simulations are conducted to discover and gain insight about the overall performance of proposed ABE coder that includes subjective (Mean Opinion Score - MOS) and objective (Perceptual Evaluation of Speech Quality - PESQ) analyses. The evaluated results for both analyses clearly advocate that proposed ABE coder outperforms legacy GSM AMR 06.90 NB coder.

A hybrid scheme of information hiding incorporating steganography as well as watermarking in the speech signal using Quantization index modulation (QIM)

This paper presents a new approach for digital Speech Steganography and watermarking in the speech signal. In the proposed approach, Quantization index modulation is applied on one of the speech signal feature called as line spectral frequency to indicate the pattern of watermark in each frame of the speech signal. Steganographic data or watermarked information is conveyed through the silence part of the speech signal. The pattern generated using QIM in line spectral frequency (LSF) of the speech gives the information about the existence of steganography or watermarking in the silence part of the speech signal. The blind detection technique is used at the receiver to detect the pattern generated at the transmitter side to identify steganography or authenticity in the speech signal. Various subjective and objective analysis are carried out on a dither modulated speech to evaluate its performance and also the robustness of the same is verified by applying a different types of communication attacks on it. The performance measures are shown in terms of set of tables and graphs. It is observed from the obtained results that, the results of different speech quality assessment parameter as well as the parameters denoting the hiding capacity of each wave file of different speakers are quite good and satisfactory.