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.
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
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.
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 ?
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
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?
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?
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
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 :)
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
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?
if i want to apply this to a real time speech and i want to cut out the empty part of the signal what should i do ... in you code its possible to cut out the beginning of the signal but i want to cut out the empty end part as well and plot the new signal with out empty values. clc close all y=audiorecorder(8000,8,1); disp('start'); %recording time 1.5 recordblocking(y,1.5); disp('end'); z=getaudiodata(y); subplot (2,1,1); plot(z); xlabel('original signal'); %sampling frequency =8000 fs = 8000; %FAME DURATION = 32ms frame_duration = 0.032; frame_len = frame_duration * fs %FRAME LENGTH = 256 n = length(z); num_frames = floor(n/frame_len) new_sig = zeros(n,1); count = 0; for k = 1:num_frames frame = z((k-1)*frame_len+1 : frame_len *k); max_val = max(frame); if (max_val>0.05); count = count + 1 ; new_sig((count-1)*frame_len+1 : frame_len*count) = frame; end end %remove the end sielent part of the speech %new_sig(80000:end) = []; subplot(2,1,2) plot (new_sig); xlabel('the new signal'); sound(new_sig,8000);
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
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 !!!
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
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....
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
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).
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.
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 :)
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
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.
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 !!!
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 !!!
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 !!!
![Eminem - Fuel (Shady Edition) feat. Westside Boogie & GRIP [Official Audio]](http://i.ytimg.com/vi/k1OquBida9s/mqdefault.jpg)
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.
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.
very nice and useful presentation sir.thank you very much
Superb imposing of sound !!!!
great
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
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
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.
Tuesander I'd say you could do it by reversing the symbol in the "if" and put zeros in the while frame. Probably
Thank you so much ! Your video is so useful for me......
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 ?
you saved my life, thank youuu
Thank you alot
IS the frame by frame analysis the same Sliding Window?
Yes Ibrahim. Its the same idea.
@@ddorran Thank you so much
Thank you for this video! It helped a lot!
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
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?
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?
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
Great tutorial, do you have any tutorials on speech analysis/synthesis using linear predictive coding (lpc)? It would be of really great help.
cheers.
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
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 :)
I don't see the benefit. Why do you think its better?
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
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?
The sampling rate will be specified in the wav file. 44100, 22050, 16000, 8000 Hz would be common for audio data.
great video!
Is it possible to define new_sig to grow with added frames instead of predefined zeros(N,1)? Like array in other programming languages
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?
hi please can you send me the code
Can you please send me your code plzzzz
Can this be done to a fourier transform as well? How would you do it?
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 :)))
That was good! thank you!!
if i want to apply this to a real time speech and i want to cut out the empty part of the signal what should i do ...
in you code its possible to cut out the beginning of the signal but i want to cut out the empty end part as well and plot the new signal with out empty values.
clc
close all
y=audiorecorder(8000,8,1);
disp('start');
%recording time 1.5
recordblocking(y,1.5);
disp('end');
z=getaudiodata(y);
subplot (2,1,1);
plot(z);
xlabel('original signal');
%sampling frequency =8000
fs = 8000;
%FAME DURATION = 32ms
frame_duration = 0.032;
frame_len = frame_duration * fs
%FRAME LENGTH = 256
n = length(z);
num_frames = floor(n/frame_len)
new_sig = zeros(n,1);
count = 0;
for k = 1:num_frames
frame = z((k-1)*frame_len+1 : frame_len *k);
max_val = max(frame);
if (max_val>0.05);
count = count + 1 ;
new_sig((count-1)*frame_len+1 : frame_len*count) = frame;
end
end
%remove the end sielent part of the speech
%new_sig(80000:end) = [];
subplot(2,1,2)
plot (new_sig);
xlabel('the new signal');
sound(new_sig,8000);
Thanks very much!
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
I see what your other comment means now. That code seems good to me if you want to store each frame.
David Dorran Thanks a lot David :)
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 !!!
hey ..
how can I find the max value for the silence frames without looking to the graph ?
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
David Dorran thank you but how can I show it as an output coz it's not workin with me !
Mona khalil Post your code and I'll take a look
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....
use audioread - it replaced wavread
Thanks so much!! You are really awesome.
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
Hi David
I would like to know, why sampling frequency was choosen as 22050(fs). Please let me know the exact reason.
Thanks.
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).
is it ok if i choose 11025 sampling frequency for environmental sounds like door knock and keyboard typing ? please explain me...
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;
sir, will u please tell me about the mfcc of a audio signal, means how to calculate the mfcc of a audio signal
Hi David,
All values of my new_sig is 0.
I think something is wrong with vector new_sig.
Can you help me?
Instead of identifying silence, how to identify the maximum amplitude/peak frame by frame? Hope to get reply from you soon. Thanks.
The following gets the maximum value in a frame:
max_val = max(frame);
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.
[max_val max_loc] = max(ip);
frame_len = 100;
seg = ip(max_loc - frame_len/2:max_loc + frame_len/2);
plot(seg)
Thank you so much and your fast response.
hi David, can i check with you how do we find the max_loc?. Thanks
where can ifind the m_file and the darryl video
when i type waveread and audioread it says undefined for both can u please tell what is problem???
i m facing this problem too? did u get solution of it ?
how to get max_value without looking the graph
thanks..
can you also tell me how to convert the new signal back into a .wav file??
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 :)
is this work published?
slvp comment je peux afficher le fram
can we use new_sig=ip(ip>.03) ?
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
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.
wavwrite(new_sig, 22050, 16, 'speech.wav')
buffer is enough
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 !!!
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 !!!
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 !!!