This paper presents a synthesis based technique to improve the listening of the humans through the device. The method that used here is sound synthesizing technique. This technique is applied to the speech synthesized from the parameters derived for the speech processing of the word “Digital Signal Processing”. From the output of the device, the more number of the signals that detected in the input were synthesized separately and filtered specifically in the sound. So the device is also being a developed mobile phone with a specific application. The mobile can do the process that what we need. It gets the input as audio in the detection circuit. The output was connected with an enhancement device to the humans.
This paper presents a synthesis based technique to improve the listening of the humans through the device. The method that used here is sound synthesizing technique. This technique is applied to the speech synthesized from the parameters derived for the speech processing of the word “Digital Signal Processing”. From the output of the device, the more number of the signals that detected in the input were synthesized separately and filtered specifically in the sound. So the device is also being a developed mobile phone with a specific application. The mobile can do the process that what we need. It gets the input as audio in the detection circuit. The output was connected with an enhancement device to the humans.
This paper presents a synthesis based technique to improve the listening of the humans through the device. The method that used here is sound synthesizing technique. This technique is applied to the speech synthesized from the parameters derived for the speech processing of the word “Digital Signal Processing”. From the output of the device, the more number of the signals that detected in the input were synthesized separately and filtered specifically in the sound. So the device is also being a developed mobile phone with a specific application. The mobile can do the process that what we need. It gets the input as audio in the detection circuit. The output was connected with an enhancement device to the humans.
This paper presents a synthesis based technique to improve the listening of the humans through the device. The method that used here is sound synthesizing technique. This technique is applied to the speech synthesized from the parameters derived for the speech processing of the word “Digital Signal Processing”. From the output of the device, the more number of the signals that detected in the input were synthesized separately and filtered specifically in the sound. So the device is also being a developed mobile phone with a specific application. The mobile can do the process that what we need. It gets the input as audio in the detection circuit. The output was connected with an enhancement device to the humans.
International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190
ICRTIET-2014 Conference Proceeding, 30th -31st August 2014
SUPER LISTENER P.MANO Electronics & communication Engg. Renganayagi Varatharaj College Engineering Sivakasi, Tamilnadu
ABSTRACT This paper presents a synthesis based technique to improve the listening of the humans through the device. The method that used here is sound synthesizing technique. This technique is applied to the speech synthesized from the parameters derived for the speech processing of the word Digital Signal Processing. From the output of the device, the more number of the signals that detected in the input were synthesized separately and filtered specifically in the sound. So the device is also being a developed mobile phone with a specific application. The mobile can do the process that what we need. It gets the input as audio in the detection circuit. The output was connected with an enhancement device to the humans.
M.DINESH Electronics & communication Engg. Renganayagi Varatharaj College Engineering Sivakasi, Tamilnadu
2. SIGNAL PROCESSING Humans are the most advanced signal processors in speech and pattern recognition, speech synthesis, so it is easy to split the human voice from the others sounds. Generally sounds are vibrations, they has compression and rarefaction.
Keywords: Digital Signal Processing, Enhancement,
Improvement of listening, mobile software, speech processing, voice signal, sound synthesis.
1. INTRODUCTION Speech processing devices like digital hearing aids, mobiles and other man machine interface in our daily life to make them more robust in noisy environmental conditions. In this technically developing world, there are lots of noise pollution were presents. So the sound proof rooms may be available at the home but one cant use everywhere. But one can easily carry their mobiles as sound proofing. Human speech can be modeled as filter acting on excitation waveform. There often occur conditions under which we measure and then transform the speech signal to another form in order to enhance our ability to communicate with each other is 4 KHz. The detected audio signal is in the form of analog signal. Then the audio was transmitted to digital form through analog to digital converter. One of the most common noises is the background noise which is present at any location. Other types of noise include channel noise which affects both digital and analog transmission. The speech signal is corrupted by various noises such as Gaussian noise, babble noise.
Most real world the signals are in analog they are
continuous in time and amplitude convert to voltage or currents using sensors and transducers analog circuits process these signals using resistors, capacitors, inductors and amplifiers, etc. analog signal processing examples audio processing in FM radios video processing in traditional TV sets. Signal processing: convert the audio wave into sequence of feature vectors. Speech recognition: Decode the sequence of feature vectors into a sequence of words.
3. METHODS 1. Filtering the specific sound method 2. Amplification method
4. AUDIO DETECTION Sound wave is a pressure wave, a detector could used to detect oscillations in pressure from a high pressure to a low pressure and back to a high pressure. As the
Divya Jyoti College of Engineering & Technology, Modinagar, Ghaziabad (U.P.), India
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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190 ICRTIET-2014 Conference Proceeding, 30th -31st August 2014
compression (High pressure) and rarefaction (Low
pressure) move through the medium, they would reach the detector 500 times per second if the frequency of the wave were 500Hz similarly if the rarefactions reach the detector 500 times/sec if the frequency of the wave 500Hz. The frequency of the sound wave not only represents a number of back and forth vibrations of the particles per unit of time, but also refers the number of compressions and rarefactions.
was shown to the user and user can take the decision. For an example, when we are in the situation there we are being hearing the sound by two men by talking to us in the same time, on that time we have to use a device that gives a advancement of listening, so we can filtered out ones sound and we also can hear the remaining ones speech clearly. This method was also used to hear small noise into amplified one. When the user is in the political meeting, the user cant hear the leaders speech clearly, if he uses a device, he can clearly hear the leaders speech.
5. FILTERING THE SPECIFIC SOUND
Algorithm
5. AMPLIFICATION METHOD The problem was explained in the above part so the solution is; the problem is to be amplified and compared then finally user can take a decision who user doesnt want. The both of their speech were identified and that was to be amplified and compared by the device then it
Algorithm I: i). S(n) --.wav file for running time=1 min. or 2 min. ii) For running time iii) Input signal iv) Original Signal ii) After applying windowing/Filter/LPC Algorithms iii) Synthesized iv) After applying codec file: codec, fs, D);
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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190 ICRTIET-2014 Conference Proceeding, 30th -31st August 2014
v) LPC=16 vi) LPC+codec vii) Input signal viii) For running time 1 sec/1min/2min. ix) Codec with file or file name (ref.16kb) x) Plot codec xi) Grid xii) Run. Algorithm II: Step1: s (n), to .wav file Step2: Input signal Step3: Original file Step4: After applying windowing/filter/LPC algorithm Step5: Synthesized Step6: after applying codec file Step7: codec, fs, D:; LPC=16+codec Step8: deburg-save and run Step9: we have to received output signal or voiced or unvoiced signal Step10: expected response, plot, original plot and synthesized plot Algorithm III: Simulation Procedure i) Entering into mat lab environment ii) Opening of the Mat Lab Software iii) After opening the Mat Lab the first screen is as iv) Shows existed data v) Selecting the project from computer as vi) Importing project into the Matlab Software as vii) Click on exe File i.e. practical values are observed
Step 8: Pitch calculation purpose is Frequency values
are decided such as high Frequency /low frequency Step 7: Unvoiced pitch can not be calculated Algorithm VI: Simulation on Real Speech Data Step 1: wavelet an 32 msec./1min.long synthesized voiced speech segment-position/dsp/ sample at 16 kHz Step 2: plot Original, synthesized adjusted high frequency components Step 3: a) segment /overlap/buffer/reshape/ the signal into 32 msec. b) decision: i) voiced, ii) unvoiced c) Pitch period is estimated d) Pre- emphasizing filtering e) The th
LPC calculated order .16 order
Step 4: a) wavelet based b) Variance of residual signal c) Variance of last four values of the wavelet coefficient for every segment or frame, frame= part of speech signal =segment=part of signal =frame d) Code vector is formed , ao = a16 ---- values =vector= direction on with
magnitude ak= ? =L, ak= ak =LPC
Simulation results from Algorithm III- Practical values: We can estimate voiced and unvoiced signal. X= 0, unvoiced signal Remaining values= voiced signals, Window=512, N=512, fs= sample frequency, window length= 16, LPC order=16 or 32, Residue= error signal, Power=0.026155, Fs= sample frequency, 1second-16000 samples, a= LPC, g=gain, = 5.4830e-004 that means 10
Algorithm IV: Step 1: s (n) =speech signal Step 2: windowing Step 3: pre-emphasizing Step 4: LPC, y (n) Algorithm: Step 1: windowing a pre-emphasizing Step 2: LPC or coefficients Step 3: wavelet based analysis Step 4: Inverse filtering Step 5: voiced/unvoiced detection (If voiced: Vowel, AEIOU, Pitch=Frequency = they are calculated as voiced signal or high magnitude Step 6: Pitchthe frequency at where the signal pick is maximum, high magnitude is Available. SignalGraph-------Energy/magnitude vs. frequency graph is identified with help of pitch values. Pitch Calculation: Magnitude or Energy vs. Frequency=PITCH
Divya Jyoti College of Engineering & Technology, Modinagar, Ghaziabad (U.P.), India
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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190 ICRTIET-2014 Conference Proceeding, 30th -31st August 2014
Original speech signal
REFERENCES [1] www.google.com [2] www.Wikipedia.com/article/frequency [3] www.Physicsclassroom.com/Audiofrequency [4] www.ECELABS.com/audiodetection [5] www.hyperphysics.com/amplifier [6] Journal of Advanced Communication Engineering- S. China Venkateswarlu
Divya Jyoti College of Engineering & Technology, Modinagar, Ghaziabad (U.P.), India
