2,472 research outputs found
Interactive Real-Time Embedded Systems Education Infused with Applied Internet Telephony
The transition from traditional circuit-switched phone systems to modern packet-based Internet telephony networks demands tools to support Voice over Internet Protocol (VoIP) development. In this paper, we introduce the XinuPhone, an integrated hardware/software approach for educating users about VoIP technology on a real-time embedded platform. We propose modular course topics for design-oriented, hands-on laboratory exercises: filter design, timing, serial communications, interrupts and resource budgeting, network transmission, and system benchmarking. Our open-source software platform encourages development and testing of new CODECs alongside existing standards, unlike similar commercial solutions. Furthermore, the supporting hardware features inexpensive, readily available components designed specifically for educational and research users on a limited budget. The XinuPhone is especially good for experimenting with design trade-offs as well as interactions between real-time software and hardware components
A Binaural Neuromorphic Auditory Sensor for FPGA: A Spike Signal Processing Approach
This paper presents a new architecture, design
flow, and field-programmable gate array (FPGA) implementation
analysis of a neuromorphic binaural auditory sensor, designed
completely in the spike domain. Unlike digital cochleae that
decompose audio signals using classical digital signal processing
techniques, the model presented in this paper processes information
directly encoded as spikes using pulse frequency modulation
and provides a set of frequency-decomposed audio information
using an address-event representation interface. In this case,
a systematic approach to design led to a generic process for
building, tuning, and implementing audio frequency decomposers
with different features, facilitating synthesis with custom features.
This allows researchers to implement their own parameterized
neuromorphic auditory systems in a low-cost FPGA in order to
study the audio processing and learning activity that takes place
in the brain. In this paper, we present a 64-channel binaural
neuromorphic auditory system implemented in a Virtex-5 FPGA
using a commercial development board. The system was excited
with a diverse set of audio signals in order to analyze its response
and characterize its features. The neuromorphic auditory system
response times and frequencies are reported. The experimental
results of the proposed system implementation with 64-channel
stereo are: a frequency range between 9.6 Hz and 14.6 kHz
(adjustable), a maximum output event rate of 2.19 Mevents/s,
a power consumption of 29.7 mW, the slices requirements
of 11 141, and a system clock frequency of 27 MHz.Ministerio de Economía y Competitividad TEC2012-37868-C04-02Junta de Andalucía P12-TIC-130
Using Transcoding for Hidden Communication in IP Telephony
The paper presents a new steganographic method for IP telephony called
TranSteg (Transcoding Steganography). Typically, in steganographic
communication it is advised for covert data to be compressed in order to limit
its size. In TranSteg it is the overt data that is compressed to make space for
the steganogram. The main innovation of TranSteg is to, for a chosen voice
stream, find a codec that will result in a similar voice quality but smaller
voice payload size than the originally selected. Then, the voice stream is
transcoded. At this step the original voice payload size is intentionally
unaltered and the change of the codec is not indicated. Instead, after placing
the transcoded voice payload, the remaining free space is filled with hidden
data. TranSteg proof of concept implementation was designed and developed. The
obtained experimental results are enclosed in this paper. They prove that the
proposed method is feasible and offers a high steganographic bandwidth.
TranSteg detection is difficult to perform when performing inspection in a
single network localisation.Comment: 17 pages, 16 figures, 4 table
Development and Evaluation of a Real-Time Framework for a Portable Assistive Hearing Device
Testing and verification of digital hearing aid devices, and the embedded software and algorithms can prove to be a challenging task especially taking into account time-to-market considerations. This thesis describes a PC based, real-time, highly configurable framework for the evaluation of audio algorithms. Implementation of audio processing algorithms on such a platform can provide hearing aid designers and manufacturers the ability to test new and existing processing techniques and collect data about their performance in real-life situations, and without the need to develop a prototype device. The platform is based on the Eurotech Catalyst development kit and the Fedora Linux OS, and it utilizes the JACK audio engine to facilitate reliable real-time performance Additionally, we demonstrate the capabilities of this platform by implementing an audio processing chain targeted at improving speech intelligibility for people suffering from auditory neuropathy. Evaluation is performed for both noisy and noise-free environments. Subjective evaluation of the results, using normal hearing listeners and an auditory neuropathy simulator, demonstrates improvement in some conditions
Delta modulation techniques for low bit-rate digital speech encoding
Includes bibliography.Two new hybrid companding delta modulators for speech encoding are presented here. These modulators differ from the Hybrid Companding Delta Modulator (HCDM) proposed by Un et al in that the two new encoders employ Song Voice Adaptation as the basis of the instantaneous compandor, rather than Constant Factor adaptation. A detailed analysis of the performance, both objective and subjective, of these hybrid codecs has been carried out. Results show that overall the two codecs developed as part of this project are better than the HCDM codec. In addition the new codecs offer simpler implementation in digital hardware than the HCDM. A Computer Aided Test (CAT) system has been developed to simplify the design and test processes for speech codecs
Low power techniques for video compression
This paper gives an overview of low-power techniques proposed in the literature for mobile multimedia and Internet applications. Exploitable aspects are discussed in the behavior of different video compression tools. These power-efficient solutions are then classified by synthesis domain and level of abstraction. As this paper is meant to be a starting point for further research in the area, a lowpower hardware & software co-design methodology is outlined in the end as a possible scenario for video-codec-on-a-chip implementations on future mobile multimedia platforms
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