I want to design a web-app for my cousin who is 2 years of age in which i have implemented a functionality in which when an image is clicked some sound gets played and the user has to make the same sound which gets recorded.
For eg-If i click on image of "Apple" the sound made is "A for Apple".Now the user has to say those words which get recorded.
Now I want to calculate the percentage of accuracy with which the user spoke.I want to know how can i know the accuracy percentage.I have not used machine learning or Natural Language Processing earlier so i want some guidance on what should i learn about or ways of implementing this functionality.I need some help on that.
Also use nodejs frameworks quite frequently so is there any module in nodejs with the help of which the above requirement can be fulfilled.
What you want to reach is a quite complex and non-trivial task that can be faced at several levels. First of all, you should answer a question in before for yourself:
What do you mean with "accuarcy"? Which metric do you want to use for that? Accuracy means to compare a result with its optimum. So what would be the optimum of saying "Apple"?
I think there are several levels on which you could measure speech accuracy:
On the audio level: Here are several correlation metrics that can compute the similarity of two audio files. See e.g. here for more details. SImply said, the idea is directly comparing the audio samples. In your case, you would need a reference audio track that is the "correct" result. The correct time alignment might become a problem though.
On the level of speech recognition: You could use a speech recognizer -- commercial or open source -- and return a string of spoken words. In this case you should think about when the recording is stopped, to limit the record length. Then you have to think about a metric that evaluates the correctness of the transcription. Some that I worked with are Levensthein-Distance or Word-Error-Rate. Wit these you can compute a similarity.
Related
My friend Prasad Raghavendra and me, were trying to experiment with Machine Learning on audio.
We were doing it to learn and to explore interesting possibilities at any upcoming get-togethers.
I decided to see how deep learning or any machine learning can be fed with certain audios rated by humans (evaluation).
To our dismay, we found that the problem had to be split to accommodate for the dimensionality of input.
So, we decided to discard vocals and assess by accompaniments with an assumption that vocals and instruments are always correlated.
We tried to look for mp3/wav to MIDI converter. Unfortunately, they were only for single instruments on SourceForge and Github and other options are paid options. (Ableton Live, Fruity Loops etc.) We decided to take this as a sub-problem.
We thought of FFT, band-pass filters and moving window to accommodate for these.
But, we are not understanding as to how we can go about splitting instruments if chords are played and there are 5-6 instruments in file.
What are the algorithms that I can look for?
My friend knows to play Keyboard. So, I will be able to get MIDI data. But, are there any data-sets meant for this?
How many instruments can these algorithms detect?
How do we split the audio? We do not have multiple audios or the mixing matrix
We were also thinking about finding out the patterns of accompaniments and using those accompaniments in real-time while singing along. I guess we will be able to think about it once we get answers to 1,2,3 and 4. (We are thinking about both Chord progressions and Markovian dynamics)
Thanks for all help!
P.S.: We also tried FFT and we are able to see some harmonics. Is it due to Sinc() in fft when rectangular wave is input in time domain? Can that be used to determine timbre?
We were able to formulate the problem roughly. But, still, we are finding it difficult to formulate the problem. If we use frequency domain for certain frequency, then the instruments are indistinguishable. A trombone playing at 440 Hz or a Guitar playing at 440 Hz would have same frequency excepting timbre. We still do not know how we can determine timbre. We decided to go by time domain by considering notes. If a note exceeds a certain octave, we would use that as a separate dimension +1 for next octave, 0 for current octave and -1 for the previous octave.
If notes are represented by letters such as 'A', 'B', 'C' etc, then the problem reduces to mixing matrices.
O = MI during training.
M is the mixing matrix that will have to be found out using the known O output and I input of MIDI file.
During prediction though, M must be replaced by a probability matrix P which would be generated using previous M matrices.
The problem reduces to Ipredicted = P-1O. The error would then be reduced to LMSE of I. We can use DNN to adjust P using back-propagation.
But, in this approach, we assume that the notes 'A','B','C' etc are known. How do we detect them instantaneously or in small duration like 0.1 seconds? Because, template matching may not work due to harmonics. Any suggestions would be much appreciated.
Splitting out the different parts is a machine learning problem all to its own. Unfortunately, you can't look at this problem in audio land only. You must consider the music.
You need to train something to understand musical patterns and progressions in the context of the type of music you give it. It needs to understand what the different instruments sound like, both mixed and not mixed. It needs to understand how these instruments are often played together, if it's going to have any chance at all at separating what's going on.
This is a very, very difficult problem.
This is a very hard problem mainly because converting audio to pitch isnt very simple due to Nyquist folding harmonics that are 22Khz+ back down and also other harmonic introductions such as saturators/distortion and other analogue equipment that introduce harmonics.
The fundamental harmonic isnt always the loudest which is why your plan will not work.
The hardest thing to measure would be a distorted guitar. The harmonic some pedals/plugins can make is crazy.
Closed. This question does not meet Stack Overflow guidelines. It is not currently accepting answers.
We don’t allow questions seeking recommendations for books, tools, software libraries, and more. You can edit the question so it can be answered with facts and citations.
Closed 6 years ago.
Improve this question
I have been searching for emotion detection through voice/speech solution on mobile (iOS) and web.
I found Moodies-iOS and Vokaturi solution, but they are not free.
I couldn't find any open source or paid version software available to integrate in my app and test the solution.
Could someone share if you have any info on this related.
Is there any OPEN SOURCE for iOS for Emotion analysis and detection through Voice/Speech, Please let me know.
As a former research in affective computing, I highly doubt you can find a ready-for-use iOS open source solution for emotion recognition from speech. The main reason is that it is a damn difficult task that requires a lot of research and a lot of proper data to train models. That is why companies like BeyondVerbal and Vokaturi do not share their models with others. Thus, you will be very lucky if you can find anything in open source, I am not even talking about iOS solutions.
I am aware about some toolkits you can use for this task (namely, the openEAR toolkit), but to build something working from it, you need an expert knowledge in the field and data to train models. A comprehensive list of databases can be found here: http://emotion-research.net/wiki/Databases. A lot of them a freely available.
As Dmytro Prylipko said it is very doubtful that there is any open-source lib for emotion recognition from speech.
You may write your own solution. It is not hard. Trouble is, as mentioned before, proper training and/or trasholding takes a lot of time and nerves.
I will give you a short theory how you should begin writing the algo, but training and so on is on you.
First big trouble is that different people differently relay their emotions vocally.
For example: one shocked person will to their shock respond with overexclaimed sentence while another will "freeze" and their response would sound very flat (almost robot-like).
Therefore you will need a lot of templates from which to learn how to classify your input speech by emotions.
You can remove some difficulties by using context recognition along with voice prosody.
That is what I'd advise you to do.
First make an algorithm that will use speech-recognized text to put it into emotion context. E.g. you can use specific words and phrases that people use when expressing different emotions.
That is easily done. You may use a neural network or simple branching or whatever.
So you will be able to recognize whether person is thankful and surprised at the same time by combining context recognition and emotions from prosody.
Now, to recognize the emotion from prosody you have to get prosody parameters and some others.
For example, some emotions may be recognized by looking at duration of particular words in a sentence.
So you have the sentence and the text of that sentence. You know that the speed of normal speech is approximately 200 words per minute. Knowing this and number of words in the sentence you can see how fast is someone talking. Then you measure the duration of each word and get its speed. By knowing how fast is the speech and how long is the word you can get normalized ratios that can be used for classifications in order to determine the closest guess of the emotion.
For instance, when someone is presented with a present that he/she likes very much, the "thank you" will sound pretty long. It will also be of higher pitch than that person's usual speech.
So the next step would be to get the average pitch for each word to see the relation between them. So you will be able to see how the sentence prosody modulates. From lower to higher, or vice versa.
Also, how prosody changes inside the phrases within the sentence.
You may go about this by comparing curves of known emotion directly, or you may use aproximation to get coefficients from the prosody curve vector. The square function does good for normal speech prosody (with no particular emotions in). So some higher order polynomial should do. So, you can get coefficients of the polynom and use them to get what emotion should whole sentence or phrase relay.
The same goes for individual words within the sentence. You get the pitch for each phoneme or syllable or just the pitch curve for e.g. every 20 ms of the word. Then you either calculate few coefficients to aproximate the polynom you decided is good enough for you, or you take the whole curve and normalize it to e.g. 30 points to use it with recognition.
To compare curves directly you may use gesture recognition algorithm by Oleg Dopertchouk:
http://www.gamedev.net/reference/articles/article2039.asp
I tried it on pitch curves of melodies, it works just fine.
The trouble is, you need a database of speech with context and emotion with clear manually done classification to give your algo something to compare with.
If you use polynomials instead of whole curves, you can do some recognition by using thresholds on coefficients, but results will be a bit shaky. Only real excuse for using coeffs at all is that you do not need to know how long is the word in question. I.e. the same polynom should work on a word with 2 phonemes and on one with 5. (should work)
You see, a theory is nice and easy. Use speech recognition, measure speech rate, and duration of each word, construct pitch curve for whole phrase and pitch curve for each word using FFT, do some comparison between ready database and the input. And walla, emotion recognized.
But where will you find the database with word curves marked with emotions.
For example, you would need for each emotion at least one pitch curve for words with different number of phonemes. At least one, because it is important whether the word starts with vowel or ends with one, or simply someone differently relays the same emotion even if the curve represents the same word.
OK, so you can say that you can make one. Where would you find recorded samples to make your curves or calculate coeffs? Hm, perhaps a recording of some drama. Not bad idea, but the acted emotions aren't the same as the natural ones.
It is a big job to teach a machine such a thing.
Oh, yeah, I almost forgot, emotions aren't only, or sometimes at all transfered using pitch changes, sometimes it's only the way in which the word is being pronounced.
So, for some cases, you would probably need LPC or some other coefficients showing some more info on how phonemes in the word sounds. Or you would need to take in view other harmonics from FFT, not just the one representing the pitch of excitation train.
The best that you can do without following my hints and developing your own algo, is to use NLTK (natural language toolkit) to develop a statistical speech (emotionally rich) model and use algorithms from there (perhaps a bit modified) to try to get to the emotion in question.
But I fear it would be a greater job than going from zero. As far as I know NLTK doesn't support emotions. Just normal speech prosody.
You may try to integrate some things I wrote about into Sphinx, to develop emotion based speech models and introduce emotion recognition directly into sphinxes VR algorithm.
If you really need this, I advise you to learn enough DSP to write your own algo, then pay someone to make you initial database from audiobooks, radio dramas and similar stuff (using a tool you provide).
After your algo starts to work reasonably well, implement autolearning by giving users an option to correct the algo's wrong guesses. After some time you will get 90% reliable algo to recognize emotions from speech.
I am trying to create an iOS app that will perform an action when it detects a clapping sound.
Things I've tried:
1) My first approach was to simply measure the overall power using an AVAudioRecorder. This worked OK but it could get set off by talking too loud, other noises, etc so I decided to take a different approach.
2) I then implemented some code that uses a FFT to get the frequency and magnitude of the live streaming audio from the microphone. I found that the clap spike generally resides in the 13kHZ-20kHZ range while most talking resides in a lot lower frequencies. I then implemented a simple thresh-hold in this frequency range, and this worked OK, but other sounds could set it off. For example, dropping a pencil on the table right next to my phone would pass this thresh-hold and be counted as a clap.
3) I then tried splitting this frequency range up into a couple hundred bins and then getting enough data where when a sound passed that thresh-hold my app would calculate the Z-Score (probability from statistics) and if the Z-Score was good, then could that as a clap. This did not work at all as some claps were not recognized and some other sounds were recognized.
Graph:
To try to help me understand how to detect claps, I made this graph in Excel (each graph has around 800 data points) and it covers the 13kHZ-21kHZ range:
Where I am now:
Even after all of this, I am still not seeing how to recognize a clap versus other sounds.
Any help is greatly appreciated!
I need to write a speech detection algorithm (not speech recognition).
At first I thought I just have to measure the microphone power and compare it to some threshold value. But the problem gets much harder once you have to take the ambient sound level into consideration (for example in a pub a simple power threshold is crossed immediately because of other people talking).
So in the second version I thought I have to measure the current power spikes against the average sound level or something like that. Coding this idea proved to be quite hairy for me, at which point I decided it might be time to research already existing solutions.
Do you know of some general algorithm description for speech detection? Existing code or library in C/C++/Objective-C is also fine, be it commercial or free.
P.S. I guess there is a difference between “speech” and “sound” recognition, with the first one only responding to frequencies close to human speech range. I’m fine with the second, simpler case.
The key phrase that you need to Google for is Voice Activity Detection (VAD) – it's implemented widely in telecomms, particularly in Acoustic Echo Cancellation (AEC).
I plan to write a conversation analysis software, which will recognize the individual speakers, their pitch and intensity. Pitch and intensity are somewhat straightforward (pitch via autocorrelation).
How would I go about recognizing individual speakers, so I can record his/her features? Will storing some heuristics for each speaker's frequencies be enough? I can assume that only one person speaks at a time (strictly non-overlapping). I can also assume that for training, each speaker can record a minute's worth of data before actual analysis.
Pitch and intensity on their own tell you nothing. You really need to analyse how pitch varies. In order to identify different speakers you need to transform the speech audio into some kind of feature space, and then make comparisons against your database of speakers in this feature space. The general term that you might want to Google for is prosody - see e.g. http://en.wikipedia.org/wiki/Prosody_(linguistics). While you're Googling you might also want to read up on speaker identification aka speaker recognition, see e.g. http://en.wikipedia.org/wiki/Speaker_identification
If you are still working on this... are you using speech-recognition on the sound input? Because Microsoft SAPI for example provides the application with a rich API for digging into the speech sound wave, which could make the speaker-recognition problem more tractable. I think you can get phoneme positions within the waveform. That would let you do power-spectrum analysis of vowels, for example, which could be used to generate features to distinguish speakers. (Before anybody starts muttering about pitch and volume, keep in mind that the formant curves come from vocal-tract shape and are fairly independent of pitch, which is vocal-cord frequency, and the relative position and relative amplitude of formants are (relatively!) independent of overall volume.) Phoneme duration in-context might also be a useful feature. Energy distribution during 'n' sounds could provide a 'nasality' feature. And so on. Just a thought. I expect to be working in this area myself.