1) Yes. frame_len (samples) = fs(samples per second) * frame_dur(seconds). If the frame_dur was 10ms and fs =22050Hz then frame_len = 220 (rounding since you can't have a non-integer number of samples). 2) To save reduce file size. You loose higher frequecies when you sample at lower frequencies. Telephony systems sample at 7kHz - listening to someone on the telephone sounds different to listening to someone on the radio. 3) Unit is Hz, but can be expressed in kHz for convenience.
@Dr.alphonceowayo3 жыл бұрын
Wow I like this video. Kind of similar to signal filtering, which I am about to apply for my research and I was just wondering how.You are a savior.
@deepaksujith9 жыл бұрын
Superb imposing of sound !!!! great
@ppsaraswathi46277 жыл бұрын
very nice and useful presentation sir.thank you very much
@TheAhmadob7 жыл бұрын
that's very good, by why did you have to breakdown the signal into frames? Why didn't you just loop over the whole length of the signal and whenever the threshold condition is met, you write that into new_sig ?
@RoadsAndTraffic6 жыл бұрын
i would like to process an audio signal using MATLAB, i should break the time series data into several segments (suitable for microcontroller memory limit) and do the signal processing step by step. could you give the idea please how to that?
@heysiri2895 жыл бұрын
you saved my life, thank youuu
@darkobul18 жыл бұрын
Is it possible to define new_sig to grow with added frames instead of predefined zeros(N,1)? Like array in other programming languages
@khinezar16297 жыл бұрын
Thank you so much ! Your video is so useful for me......
@kwakuowusuboateng43994 жыл бұрын
Great tutorial David. Please, can you help me with a plot of Short Term Energy (STE) of signal and number of frames. Also, please how do you show e.g. 10 frames with all signals showing in a single plot
@matinkaviani8014 жыл бұрын
Hi, thank you for your informative video. As I'm new to audio processing, can I ask why You took 22050 as the sampling rate?
@ddorran4 жыл бұрын
The sampling rate will be specified in the wav file. 44100, 22050, 16000, 8000 Hz would be common for audio data.
@musadiness110 жыл бұрын
thank you for the tutorial! I think you can optimize like this with 2 for cycles instead of removing the silence at the end: count = 0; for k=1 : number_frames frame = x( (k-1)*frame_len +1 : frame_len*k); max_val = max(frame); if (max_val>0.04) count = count +1; end end new_signal = (count) %creation of the new signal with the proper size for k=1 : number_frames frame = x( (k-1)*frame_len +1 : frame_len*k); max_val = max(frame); if (max_val>0.04) count = count +1; new_signal((count-1)*frame_len +1 : frame_len*count)=frame; end end
@ibrahimallafi14603 жыл бұрын
Thank you alot IS the frame by frame analysis the same Sliding Window?
@ddorran3 жыл бұрын
Yes Ibrahim. Its the same idea.
@ibrahimallafi14603 жыл бұрын
@@ddorran Thank you so much
@xyz88697 жыл бұрын
sir, kindly answer to this question: how can we separate silent speech on the basis of energy of each frame instead of calculating the maximum amplitude of each frame?
@russtyglifonea284011 жыл бұрын
hello sir!!!... how can i extract the green channel in a pixel processing? can i get the sample of a codes of how to extract?thanks :)))
@Hana-cv7ze10 жыл бұрын
Thank you for this video! It helped a lot!
@meenas27547 жыл бұрын
Greetings sir, can I ask you a question? What part of audio is non sensitive?? That means if we change some values in audio it should not affect that audio.. which part can I change? Thanks in advance
@kewlaldo104 жыл бұрын
Can this be done to a fourier transform as well? How would you do it?
@yehiaelyamani69437 жыл бұрын
great video!
@MrArammos10 жыл бұрын
I think it should be better to write it as follows: for i = 1:num_frames frame(:, i) = x( (i - 1) * frame_len + 1 : frame_len*i); end Excuse me if i'm wrong i'm new to Matlab and signal processing :)
@ddorran10 жыл бұрын
I don't see the benefit. Why do you think its better?
@rohanpatil252910 жыл бұрын
Hi David I would like to know, why sampling frequency was choosen as 22050(fs). Please let me know the exact reason. Thanks.
@ddorran10 жыл бұрын
Its a common sampling rate used to capture audio signals. While we can hear up to about 22kHz most of the "useful" auditory information is in the lower frequencies (from an intellibility perspective). This is why CD quality is 44kHz but also why we can get away with using lower sampling rates (like 8kHz for telephony).
@khinezar16297 жыл бұрын
is it ok if i choose 11025 sampling frequency for environmental sounds like door knock and keyboard typing ? please explain me...
@MrArammos10 жыл бұрын
I have a question on how to break our signal into frames, i'm running your script and frame is a 1xframe_len vector. I think that the way you wrote it extract only the last frame. We want for our frame to be a num_frames x frame_length matrix not a vector i think
@ddorran10 жыл бұрын
I see what your other comment means now. That code seems good to me if you want to store each frame.
@MrArammos10 жыл бұрын
David Dorran Thanks a lot David :)
@KansasFashion6 жыл бұрын
Why I couldn't use wavread. When I type ip = wavread('file'). It says that wavread is undefined and when I type help wavread, it doesn't work at all....
@ddorran6 жыл бұрын
use audioread - it replaced wavread
@KansasFashion6 жыл бұрын
Thanks so much!! You are really awesome.
@Tuesander8 жыл бұрын
Hi David, amazing vids. You are saving my life one video at a time. One question: If I want to gate the silence instead of removing it how do I go about it? Thinking it should be simple since the "for loop" already has set the max_val in each frame. Kind regards
@ddorran8 жыл бұрын
I'm afraid I don't understand what you mean by gating the silence. If you can clarify and upload an attempt I'll take a look
@Tuesander8 жыл бұрын
Yes, I mean a gate like the plugins used in digital audio workstations. Where you set a threshold and don't allow audio below that threshold to pass through. Basically turning down the volume when there is pauses instead of removing it.
@janm.44967 жыл бұрын
Tuesander I'd say you could do it by reversing the symbol in the "if" and put zeros in the while frame. Probably
@beantish11 жыл бұрын
Great tutorial, do you have any tutorials on speech analysis/synthesis using linear predictive coding (lpc)? It would be of really great help. cheers.
@Error_.4043 жыл бұрын
hey hello I know it was 7 years ago but did you find something about coding LPC because this is exactly what I have to do for ma project pleaase
@monakhalil89229 жыл бұрын
hey .. how can I find the max value for the silence frames without looking to the graph ?
@ddorran9 жыл бұрын
The following gets the maximum value in a frame: max_val = max(frame) put this line of code inside the IF statement to find the max of each 'silent' frame
@monakhalil89229 жыл бұрын
David Dorran thank you but how can I show it as an output coz it's not workin with me !
@ddorran9 жыл бұрын
Mona khalil Post your code and I'll take a look
@DiegoCarvacho7 жыл бұрын
That was good! thank you!!
@brendanvanbiljon38619 жыл бұрын
Hi great tutorial. I used this code to count from 1 to 10. Can you please show me how to extract each word separately and the play each number one by one?
@entretaiment71847 жыл бұрын
hi please can you send me the code
@yasminesalahi50227 жыл бұрын
Can you please send me your code plzzzz
@derrickp134511 жыл бұрын
Please could you tell me what this does in the program count = count + 1; new_sig((count-1)*frame_len+1:frame_len*count) = frame;
@babamusaabbagoni363811 жыл бұрын
Thanks very much!
@shinygene848610 жыл бұрын
Instead of identifying silence, how to identify the maximum amplitude/peak frame by frame? Hope to get reply from you soon. Thanks.
@ddorran10 жыл бұрын
The following gets the maximum value in a frame: max_val = max(frame);
@shinygene848610 жыл бұрын
I see. Let's say the graph is plot against amplitude and time(in seconds), how to specify the time (automatically without zooming in to find the time) where the max-value is? Thank you so much for your help.
hi David, can i check with you how do we find the max_loc?. Thanks
@vickysteev89648 жыл бұрын
Hi, i am Vicki Right now i am learning matlab, but i am just begginer at matlab, right now i want to learn about how to using low and high pass filter, Savitzky-Golay Filtering (SGOLAYFILT), mean and varians, and also find out the peak and valley point from the gyro and accelerometer signal, and all my data i get from excel. can You give me some tutorial or code how to make it I really hope You can help me to figure it out Best Regards Vicki
@rock95raw336 жыл бұрын
thanks for the great stuff man seriously WOWs!! :)...i'd like to ask u how to detect a tone embedded in speech using matlab ???? plz help man what to do exactly i dont know !!!
@AdityaSingh-yx3ky11 жыл бұрын
sir, will u please tell me about the mfcc of a audio signal, means how to calculate the mfcc of a audio signal
@MrNextboy11 жыл бұрын
Hi David, All values of my new_sig is 0. I think something is wrong with vector new_sig. Can you help me?
@TopVideos-ts2cp6 жыл бұрын
when i type waveread and audioread it says undefined for both can u please tell what is problem???
@naveenchawla705 жыл бұрын
i m facing this problem too? did u get solution of it ?
@sarahramadan25649 жыл бұрын
where can ifind the m_file and the darryl video
@mariemabdelhaye10797 жыл бұрын
slvp comment je peux afficher le fram
@princessgenes11 жыл бұрын
can you also tell me how to convert the new signal back into a .wav file??
@BHARGAB100010 жыл бұрын
thanks..
@MuhammadAsim-mh4xo8 жыл бұрын
is this work published?
@saranyagopal91319 жыл бұрын
how to get max_value without looking the graph
@ParthRajSingh10 жыл бұрын
can we use new_sig=ip(ip>.03) ?
@ddorran10 жыл бұрын
Not exactly sure what you are trying to do but I think new_sig = ip(find(ip>0.03)) might be what you are looking for. This will just create a new_sig variable which contains all samples greater than 0.03
@ParthRajSingh10 жыл бұрын
Yes i was trying to do same as new_sig = ip(find(ip>0.03)) , it works without 'find' method. Basically I am looking for the optimum frame length for assuming sound signal as wide sense stationary to calculate cross correlation.
@adembagi10 жыл бұрын
hello mr dorran firstly ı want to thanks you so much for your sharing this videos help to me so much have you fır filter codes for speech recognition thanks again you are good man :)
@ddorran11 жыл бұрын
wavwrite(new_sig, 22050, 16, 'speech.wav')
@sbkyoutu8 жыл бұрын
buffer is enough
@rock95raw336 жыл бұрын
thanks for the great stuff man seriously WOWs!! :)...i'd like to ask u how to detect a tone embedded in speech using matlab ???? plz help man what to do exactly i dont know !!!
@rock95raw336 жыл бұрын
thanks for the great stuff man seriously WOWs!! :)...i'd like to ask u how to detect a tone embedded in speech using matlab ???? plz help man what to do exactly i dont know !!!
@rock95raw336 жыл бұрын
thanks for the great stuff man seriously WOWs!! :)...i'd like to ask u how to detect a tone embedded in speech using matlab ???? plz help man what to do exactly i dont know !!!