264 research outputs found
Using IEEE 802.15.4/ZigBee in audio applications
Most of the current uses for ZigBee and IEEE 802.15.4 focus on control applications. However, there are other areas that will benefit from the standardisation, low cost and possibly low power of ZigBee/IEEE 802.15.4. This paper focuses on the use of ZigBee/IEEE802.15.4 for audio applications. We will discuss the advantages and theoretical limits of ZigBee/IEEE 802.15.4 for this kind of applications. We will then present a design that we used as starting point to develop applications related to the transfer of audio data
A Practical Application of Wave-Pipelining Theory on a Adaptive Differential Pulse Code Modulation Coder-Decoder Design
Pipeline architectures are often considered in VLSI designs that require high throughput. The draw-backs for traditional pipelined architectures are the increased area, power, and latency required for implementation. However, with increased design effort, wave-pipelining can be applied as an alternative to a pipelined circuit to reduce the pipeline area, power, and latency while maintaining the original functionality and timing of the overall circuit. The objective of this paper is the successful application of the theories of wave-pipelining in a practical digital system. To accomplish this, the pipelined portion of an Multi-Channel Adaptive Differential Pulse Code Modulation (ADPCM) Coder-Decoder (CODEC) is replaced with a wave pipeline design
The voice activity detection (VAD) recorder and VAD network recorder : a thesis presented in partial fulfilment of the requirements for the degree of Master of Science in Computer Science at Massey University
The project is to provide a feasibility study for the AudioGraph tool, focusing on two application areas: the VAD (voice activity detector) recorder and the VAD network recorder. The first one achieves a low bit-rate speech recording on the fly, using a GSM compression coder with a simple VAD algorithm; and the second one provides two-way speech over IP, fulfilling echo cancellation with a simplex channel. The latter is required for implementing a synchronous AudioGraph. In the first chapter we introduce the background of this project, specifically, the VoIP technology, the AudioGraph tool, and the VAD algorithms. We also discuss the problems set for this project. The second chapter presents all the relevant techniques in detail, including sound representation, speech-coding schemes, sound file formats, PowerPlant and Macintosh programming issues, and the simple VAD algorithm we have developed. The third chapter discusses the implementation issues, including the systems' objective, architecture, the problems encountered and solutions used. The fourth chapter illustrates the results of the two applications. The user documentations for the applications are given, and after that, we analyse the parameters based on the results. We also present the default settings of the parameters, which could be used in the AudioGraph system. The last chapter provides conclusions and future work
Comparison between Two Algorithms of 32kb/s ADPCM using QAM Signal at 16.8kb/s
This paper studies theĀ comparison between two algorithms of 32kb/s ADPCM systems. The first algorithm uses 4-bit quantizer with sampling rate of 8000 sample/sec, and the second algorithm uses 5-bit quantizer with sampling rate of 6400 sample/sec. The comparison is done using QAM signal at data rate of 16.8kb/s. Two models of QAM signals are used, the first model operates at symbol rate of 2400 baudĀ Ā with each symbol is represented by 7-bit, while, the second model operates at symbol rate of 2800 baud with each symbol is represented by 6-bit. The contribution of this paper is that sending the second model of QAM signal over ADPCM with 5-bit quantizer. Simulation results show that the performance of ADPCM with 5-bit quantizer is better than 4-bit quantizer for both models of QAM signals. Also, the performance of ADPCM using second model is better than the first model. Furthermore, the performance with circular constellation is better than rectangular on
New Directions in Subband Coding
Two very different subband coders are described. The first is a modified dynamic bit-allocation-subband coder (D-SBC) designed for variable rate coding situations and easily adaptable to noisy channel environments. It can operate at rates as low as 12 kb/s and still give good quality speech. The second coder is a 16-kb/s waveform coder, based on a combination of subband coding and vector quantization (VQ-SBC). The key feature of this coder is its short coding delay, which makes it suitable for real-time communication networks. The speech quality of both coders has been enhanced by adaptive postfiltering. The coders have been implemented on a single AT&T DSP32 signal processo
Differential encoding techniques applied to speech signals
The increasing use of digital communication systems has
produced a continuous search for efficient methods of speech
encoding.
This thesis describes investigations of novel differential
encoding systems. Initially Linear First Order DPCM systems
employing a simple delayed encoding algorithm are examined.
The systems detect an overload condition in the encoder, and
through a simple algorithm reduce the overload noise at the
expense of some increase in the quantization (granular) noise.
The signal-to-noise ratio (snr) performance of such d codec has
1 to 2 dB's advantage compared to the First Order Linear DPCM
system.
In order to obtain a large improvement in snr the high
correlation between successive pitch periods as well as the
correlation between successive samples in the voiced speech
waveform is exploited. A system called "Pitch Synchronous
First Order DPCM" (PSFOD) has been developed. Here the difference
Sequence formed between the samples of the input sequence in the
current pitch period and the samples of the stored decoded
sequence from the previous pitch period are encoded. This
difference sequence has a smaller dynamic range than the original
input speech sequence enabling a quantizer with better resolution
to be used for the same transmission bit rate. The snr is increased
by 6 dB compared with the peak snr of a First Order DPCM codea.
A development of the PSFOD system called a Pitch Synchronous
Differential Predictive Encoding system (PSDPE) is next investigated.
The principle of its operation is to predict the next sample in
the voiced-speech waveform, and form the prediction error which
is then subtracted from the corresponding decoded prediction
error in the previous pitch period. The difference is then
encoded and transmitted. The improvement in snr is approximately
8 dB compared to an ADPCM codea, when the PSDPE system uses an
adaptive PCM encoder. The snr of the system increases further
when the efficiency of the predictors used improve. However,
the performance of a predictor in any differential system is
closely related to the quantizer used. The better the quantization
the more information is available to the predictor and the better
the prediction of the incoming speech samples. This leads
automatically to the investigation in techniques of efficient
quantization. A novel adaptive quantization technique called
Dynamic Ratio quantizer (DRQ) is then considered and its theory
presented. The quantizer uses an adaptive non-linear element
which transforms the input samples of any amplitude to samples
within a defined amplitude range. A fixed uniform quantizer
quantizes the transformed signal. The snr for this quantizer
is almost constant over a range of input power limited in practice
by the dynamia range of the adaptive non-linear element, and it
is 2 to 3 dB's better than the snr of a One Word Memory adaptive
quantizer.
Digital computer simulation techniques have been used widely
in the above investigations and provide the necessary experimental
flexibility. Their use is described in the text
Non-intrusive identification of speech codecs in digital audio signals
Speech compression has become an integral component in all modern telecommunications networks. Numerous codecs have been developed and deployed for efficiently transmitting voice signals while maintaining high perceptual quality. Because of the diversity of speech codecs used by different carriers and networks, the ability to distinguish between different codecs lends itself to a wide variety of practical applications, including determining call provenance, enhancing network diagnostic metrics, and improving automated speaker recognition. However, few research efforts have attempted to provide a methodology for identifying amongst speech codecs in an audio signal. In this research, we demonstrate a novel approach for accurately determining the presence of several contemporary speech codecs in a non-intrusive manner. The methodology developed in this research demonstrates techniques for analyzing an audio signal such that the subtle noise components introduced by the codec processing are accentuated while most of the original speech content is eliminated. Using these techniques, an audio signal may be profiled to gather a set of values that effectively characterize the codec present in the signal. This procedure is first applied to a large data set of audio signals from known codecs to develop a set of trained profiles. Thereafter, signals from unknown codecs may be similarly profiled, and the profiles compared to each of the known training profiles in order to decide which codec is the best match with the unknown signal. Overall, the proposed strategy generates extremely favorable results, with codecs being identified correctly in nearly 95% of all test signals. In addition, the profiling process is shown to require a very short analysis length of less than 4 seconds of audio to achieve these results. Both the identification rate and the small analysis window represent dramatic improvements over previous efforts in speech codec identification
A New covert channel over RTP
In this thesis, we designed and implemented a new covert channel over the RTP protocol. The covert channel modifies the timestamp value in the RTP header to send its secret messages. The high frequency of RTP packets allows for a high bitrate covert channel, theoretically up to 350 bps. The broad use of RTP for multimedia applications, including VoIP, provides plentiful opportunities to use this channel. By using the RTP header, many of the challenges present for covert channels using the RTP payload are avoided. Using the reference implementation of this covert channel, bitrates of up to 325 bps were observed. Speed decreases on less reliable networks, though message delivery was flawless with up to 1% RTP packet loss. The channel is very difficult to detect due to expected variations in the timestamp field and the flexible nature of RTP
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