Speech Compression - a novel method
#1
Music 

Speech Compression - a novel method

This paper illustrates a novel method of speech compression and transmission. This method saves the transmission bandwidth required for the speech signal by a considerable amount. This scheme exploits the property of low pass nature of the speech signal. Also this method applies equally well for any signal, which is low pass in nature, speech being the more widely used in Real Time Communication, is highlighted here.

As per this method, the low pass signal (speech) at the transmitter is divided into set of packets, each containing, say N number of samples. Of the N samples per packet, only certain lesser number of samples, say N alone are transmitted. Here is less than unity, so compression is achieved. The N samples per packet are subjected to a N-Point DFT. Since low pass signals alone are considered here, the number of significant values in the set of DFT samples is very limited. Transmitting these significant samples alone would suffice for reliable transmission. The number of samples, which are transmitted, is determined by the parameter .

The parameter is almost independent of the source of the speech signal. In other methods of speech compression, the specific characteristics of the source such as pitch are important for the algorithm to work. An exact reverse process at the receiver reconstructs the samples. At the receiver, the N-point IDFT of the received signal is performed after necessary zero padding. Zero padding is necessary because at the transmitter of the N samples only N samples are transmitted, but at the receiver N samples are again needed to honestly reconstruct the signal.

Hence this method is efficient as only a portion of the total number of samples is transmitted thereby saving the bandwidth. Since the frequency samples are transmitted the phase information has also to be transmitted. Here again by exploiting the property of signals and their spectra that the PHASE INFORMATION CAN BE EMBEDDED WITHIN THE MAGNITUDE SPECTRUM by using simple mathematics without any heavy computations or by increasing the bandwidth.

Also the simulation result of this method shows that smaller the size of the packet the more faithful is the reproduction of received signal that is again an advantage as the computation time is reduced. The reduction in the computation time is due to the fact that the transmitter has to wait until N samples are obtained before starting the transmission. If N is small, the transmitter has to wait for a less duration of time and a smaller value of N achieves a better reconstruction at the receiver.

Thus this scheme provides a more efficient method of speech compression and this scheme is also very easy to implement with the help of available high-speed processors.Transmitting the spectrum of the signal instead of transmitting the original signal is far more efficient. This is because the energy of the speech signal above 4 kHz is negligible; we can very well compute the spectrum of the signal and transmit only the samples that correspond to 4 KHz of the spectrum irrespective of the sampling frequency. By this type of transmission we can save the bandwidth required for transmission considerably. Also it is not necessary that we have to transmit all the samples corresponding to the 4 kHz frequency as it is sufficient to transmit a fraction of the samples without any degradation in the quality.

Since the spectrum is considered in the above method both the magnitude and phase information must be transmitted to reproduce the signal without any error. But this requires twice the actual bandwidth. Exploiting the property of real and even signals can solve this problem. The spectrum of the samples is real and evenliness is artificially introduced such that their spectra are also real and even. Thus by simple mathematics the complete phase information is embedded within the magnitude spectrum and it is needed only to send 'aN' samples instead of '2N'samples of the spectra (Magnitude and phase).

Adopting all these procedures and embedding the phase information in the magnitude spectrum have performed a MATLAB simulation performed to determine the optimum value of 'a' and 'N'. The result of the simulation is also provided.
Reply

Important Note..!

If you are not satisfied with above reply ,..Please

ASK HERE

So that we will collect data for you and will made reply to the request....OR try below "QUICK REPLY" box to add a reply to this page
Popular Searches: speech compression a novel method ppt, advantages of co precipitation method, distflow method, speech compression using gsm rpe ltp, what is taguchi method, adpcm for speech compression pdf, speech compression in dsp,

[-]
Quick Reply
Message
Type your reply to this message here.

Image Verification
Please enter the text contained within the image into the text box below it. This process is used to prevent automated spam bots.
Image Verification
(case insensitive)

Possibly Related Threads...
Thread Author Replies Views Last Post
  A NOVEL METHOD OF COMPRESSING SPEECH WITH HIGHER BANDWIDTH EFFICIENCY seminar surveyer 5 2,309 02-04-2015, 04:28 PM
Last Post: seminar report asees
Music Adaptive Blind Noise Suppression in some Speech Processing Applications Computer Science Clay 5 5,056 26-07-2013, 02:37 PM
Last Post: computer topic
  APPLE – A Novel Approach for Direct Energy Weapon Control project topics 13 6,990 04-03-2013, 11:43 AM
Last Post: seminar details
  Artificial intelligence for speech recognition computer science crazy 1 2,145 26-11-2012, 02:14 PM
Last Post: seminar details
  Fractal Antennas: A Novel Miniaturization Technique for Wireless Communications seminar presentation 3 3,640 20-11-2012, 01:17 PM
Last Post: seminar details
  COMMAND BY SPEECH RECOGNITION computer girl 1 1,414 27-10-2012, 01:33 PM
Last Post: seminar details
  MPEG-2 VIDEO COMPRESSION TECHNIQUE full report project topics 4 5,619 24-10-2012, 12:21 PM
Last Post: Govind Mohan
  Histogram Specification: A Fast and Flexible Method to Process Digital Images computer girl 2 1,548 20-10-2012, 01:27 PM
Last Post: seminar details
  Seminar Report on Audio Compression computer girl 0 928 11-06-2012, 02:14 PM
Last Post: computer girl
  A NEW ITERATIVE SPEECH ENHANCEMENT SCHEME BASED ON KALMAN FILTERING computer girl 0 1,056 05-06-2012, 11:25 AM
Last Post: computer girl

Forum Jump: