Sapna George

New implementation techniques of an efficient MPEG advanced audio coder

MPEG-AAC is the current state of the art in audio compression technology. The CD-quality promised at bit rate as low as 64 kbps makes AAC a strong candidate for high quality low bandwidth audio streaming applications over wireless network. Besides this low bit rate requirement, the codec must be able to run on personal wireless handheld devices with its inherent low power characteristics. While the AAC standard is definite enough to ensure that a valid AAC stream is correctly decodable by all AAC decoders, it is flexible enough to accommodate variations in implementation, suited to different resources available and application areas. This paper reviews various implementation techniques of the encoder. We then proposed our method of an optimized software implementation of MPEG-AAC (LC profile). The coder is able to perform encoding task using half the processing power compared to standard implementation without significant degradation in quality as shown by both subjective listening test and an ITU-R compliant quality-testing program (OPERA).

Investigation of different frequency estimation techniques using the phase vocoder

Sinusoidal models have long been used in the analysis and synthesis of audio signals. Many of these models including the vocoder choose critical frequencies in the amplitude spectrum for reconstruction in the synthesis section. In these models, it is important to estimate the true frequency of the signal. In this paper, we investigate the performance of traditional methods of frequency estimation using the phase vocoder. The performance of the frequency estimation techniques under varying signal to noise ratio (SNR) and different windows is analyzed. It will be shown that frequency estimation via the peak detection of the discrete Fourier transform (DFT) performs better at high SNR. We also study the effect of the presence of multiple sinusoids in the signal on the frequency estimation techniques. The mathematical analysis included is rigorous but has been presented in a simplified manner for easy interpretation.

Delay coefficients based variable step size algorithm for subband affine projection adaptive filters

Subband adaptive filters are preferred in acoustic echo cancellation systems with long echo tail lengths due to speed of convergence and complexity savings. Recently, a new and novel subband affine projection (SAP) algorithm was reported based on the polyphase decomposition of the adaptive filter and noble identities. For good system performance it is important to have a good variable step size (VSS) algorithm as part of an adaptive filter. In this paper, based on the method of delay coefficients (DC), we propose1 a VSS algorithm for the SAP adaptive filter, which is called as delay coefficients based variable step size subband affine projection algorithm (DC-VSS-SAP). We examine in detail the similarities and differences between DC method for the subband and fullband scenarios. Further, we show how the method of DC can be used to detect changes in echo paths and speed up convergence of the adaptive filter.

Dual channel phase flag determination for coupling bands in a transform coder for high quality audio

A method and apparatus for subband phase flag determination for coupling of channels in a dual channel audio encoder is based on a psychoacoustic model of the human auditory system. The method and apparatus are applicable to audio encoders which utilize a coupling channel to combine certain frequency components of the input audio signals. The method ensures a least square error between the original channel frequency coefficients at the encoder and the estimated coefficients at the decoder by determining the sign of the dot product of the coefficients for one of the channels and the coupling coefficients. No restriction is placed on the strategy utilized for generating the coupling channel coefficients or the coupling coordinates.

Comments on "Variable Explicit Regularization in Affine Projection Algorithm: Robustness Issues and Optimal Choice

The above paper presented a variable regularized affine projection algorithm (VR-APA). A delay coefficients-based variable regularized affine algorithm (DC-VR-APA) was presented in [2], [3]. It was shown in [1, pp. 2103, (39)] that DC-VR-APA is an equivalent and alternate manifestation of the VR-APA when the affine projection order is equal to 1. However, results pertaining to input speech signals and presence of near-end speech activity 1 were not reported in. For the same set of parameters, this letter 2 shows that DC-VR-APA is more robust to the above conditions than the VR-APA. Motivated by some of the explanations in, we present the reasons for the robustness of DC-VR-APA in the above conditions.

An audio signal scaling technique harmonic grouping and shifting

In this paper, we enumerate the specific advantages of using a sinusoidal representation for scaling of audio signals. Most existing systems scale signals using time domain algorithms. Here, we discuss a frequency domain technique to scale signals by any desired factor. We also compare the computational savings of the proposed algorithm with traditional time domain methods.

Personalized voice command systems in multi modal user interface

The goal of this paper is to describe the voice command system as part of the multi modal user interface for residential application project demoed at CES 2012. The application is a 3D TV panel which can be controlled through face recognition, gesture, and speech. The speech interface is invoked using activation keyword, and terminated in similar fashion with de-activation keyword. Speaker recognition is performed on the activation keyword to allow personalization of the voice commands available to the particular user, who in this scenario is a member of the household. A separate setting is also devised to enable guest user to have basic interaction with the system. Template matching scheme using dynamic time warping is employed for its simplicity and robustness to noise. The template chosen is a cluster of Gaussian Mixture Model (GMM), each representing a sub-word unit. A state model for voice interaction is presented to allow efficient operation of this interface.

Low bitrate MPEG 1 layer III encoder

Compression algorithms have a constant tradeoff between higher compression ratios at the cost of better quality. The number of bits assigned in the standard MPEG encoders is controlled by the signal to masking thresholds and the scalefactor calculations performed in the psychoacoustic model of the algorithm. The developed algorithm assigns lower bits to audio samples without significant degradation in quality.