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Questions tagged [speech]

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1answer
52 views

Comparing signals recorded using diffrent methods

Suppose we have two signals( focus to audio) recorded with different methods, or more simply, suppose we have two audio signals recorded with different microphone, and we want to confirm that these ...
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1answer
101 views

Establishing Phrase Frequency in Audio Files

Is there any Application, Programming Library or Theory that could be used to identify human language phrases in a set of audio files, so as to calculate the frequency of occurrence of each such ...
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1answer
462 views

MFCC window size at different sampling rates

The general recommendation for window size when calculating MFCC seems to be 20-40 msec. This is most often recommended in a context of 16000 samples per second, so leading to a window containing 320-...
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1answer
55 views

Is there a method to combine speech signals recorded by different microphones?

I have 3 speech files of the same speech recorded by 3 different microphones at the same time. I'm only concerned to improve the speech quality (in terms of SNR and intelligibility) by combining the ...
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1answer
674 views

Is Cross correlation between stationary random signal and noise zero for all lags?

If $x(n)$ is a stationary random process (let say small 20ms window of speech which can be assumed stationary) and $v(n)$ is said to be uncorrelated noise, can we say that cross correlation of $x(n)$ ...
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1answer
65 views

Why are the observation features of an HMM-based recognition/synthesis system modeled by a Gaussian distribution?

Why are the observation features (namely MFCCs) of an HMM-based recognition/synthesis system modeled by a Gaussian distribution? Even the state duration is modeled by a Gaussian in this paper: K. ...
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1answer
107 views

What are good estimates for phonemes length?

I am working in a phoneme recognition project, the language that I am using is "English". I have a tool that converts each written phrase in phonemes. Also I have an estimation of the time length of ...
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1answer
910 views

Removing vocoder effect from audio file

There are many tutorials and guides explaining how to create a "robotic" voice by using a vocoder or a mix of filters that change pitch, tempo and maybe add a slight echoed delay. However, is it ...
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2answers
355 views

PSOLA how to calculate synthesized Marks

I have successfully extracts all Pitch Markers from waveforms, I used autocorrelation to find my pitch periods, The markers can be observed in the figure below: I want change my signal (Pitch or ...
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1answer
309 views

Speech recognition library for detecting phrase in file

I want to find all occurrences of a specific phrase in a set of audio files (it's a catch-phrase in a TV show). The phrase is pronounced by different people with different speed etc. What is the best ...
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1answer
168 views

Mel Cepstral Distortion

I am working on a speech synthesis model and I am looking to evaluate my synthesized speech. I found that most people use the Mel Cepstral Distortion (MCD) which can be calculated by the formula: $$\...
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1answer
275 views

autoregressive moving average code implementation

I am new to DSP and i am trying to take a wav (human speech) file and apply ARMA filter and plot its PSD graph in python. I see that there are a lot of AR implementations but almost none ARMA. I ...
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0answers
58 views

peak-to-peak test between two signal

Suppose that we have different signals (Pulses). I would like to study which signal has deterministic peak-to-peak distant (lag). For example A=[1 0 0 1 0 0 0 1 0 0 1] and B=[1 0 0 1 0 0 1 0 0 1] and ...
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0answers
36 views

Examples of cases where “invariance to transformation” on signal is important

I realize that this might not be the most well-formulated question, but I figured I'd throw it out there as an open-ended discussion that could turn up some interesting input. I'm relatively new to ...
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1answer
96 views

How best to digitize a noisy phone recording for filtering?

How do I digitize an analog telephone recording and filter it with a Chebyshev filter?
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1answer
174 views

Speech Recognition - Subtitles

I am looking for ANY advice anyone can give me regarding on this problem. I'm looking to start a project on subtitles, basically, if we take a Youtube video of a breaking news story, subtitles are ...
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1answer
721 views

Averaging Different Length Speech Signal

I have around 10 wav audio files all containing same word spoken i.e. 'Hello' but all having different length. So lets say 1st file when loaded consist of 20000 samples, 2nd of 15000, 3rd of 18000 ...
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2answers
1k views

Speech Analysis - Zero-Crossing

Basically, (I know I've asked a simular question, but I have been away and really thought about it this time)! Let me try and explain this in the best way possible: Basically, I'm attempting to ...
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2answers
40 views

Converting speech audio to telephone audio

I am trying to train a machine learning algorithm on telephony speech audio. However, there isn't really enough data for this anywhere that I can find. My solution is to just use speech audio from ...
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1answer
205 views

Non human formant frequencies

Some animals are specially trained to repeat human phrases, others just make sounds that people interpret as speech. In this video a rooster seems to say "Cristo Viene" (Christ is coming). Taking ...
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1answer
2k views

what is probability density function (PDF) of periodic signal

anybody knows what is the best pdf for periodic signal ? I have signal with 480 samples (16000*0.03=480) but I could not find the exact pdf for that. sometimes it appear like 2 mixed Gaussian.
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1answer
878 views

Detect When Someone Has Stopped Talking

What's an extremely efficient way of detecting when a user has stopped talking into a microphone? When I use systems like Siri it can detect when I've stopped talking almost immediately, even when ...
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1answer
1k views
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3answers
2k views

how can i center an audio signal?

I'm trying to reconstruct an audio signal, and below is what i got after rescaling. How can i obtain centered audio signal such as below? SOLVED: ruoho ruotsi's answer addresses my problem :) i ...
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1answer
930 views

The logarithm Energy measure in Speech

I am studying the mel-frequency cepstral coefficient (MFCC) and have downloaded a pdf which is related to MFCC from this link. It is about the "ETSI ES 201 108" ETSI ES 201 108 - Speech ...
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1answer
2k views

How to compare vocal words in matlab

We as a group are new to signal processing and speech recognition. We have stored .wav sounds of three different words and we are trying to record a new .wav sound and have the system compare the ...
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1answer
238 views

Audio/speech detection, verification

I would like to write a program to compare two '.wav' files, one of which is the original played file with human speech and the other is a recorded '.wav' file. The program I wish to write in python ...
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1answer
22 views

Characterization of the amount of modulation in a signal

We have used the ICRA babble noise to generate an 8-talker babble. As a background - babble noise is used as a substitute for competing talkers as babble noise lacks lexical content, but contains the ...
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1answer
80 views

Frequency detection in audio and what parameter or set of parameters help/limit frequency detection

I'm trying to learn audio and voice/speech processing techniques out of interest. I intend to learn, experiment with algorithms and design, implementation of such systems, frameworks using programming ...
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1answer
124 views

Trying to compare two speech samples that have been recorded by smartphones in pockets

I'm trying to compare two audio samples of speech from two different Android smartphones to determine if the people are co-located. I'm doing this using the frequency bins of an FFT between 300 Hz - ...
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1answer
242 views

Raw speech data into spectrogram

I am using Delphi trying to make a very simple speech recognition system. The system collects raw data well, but I have a hard time turning the data into a spectrogram: The system collects an array ...
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1answer
61 views

Fine-grained evaluation of speech recognition results

I know that usually ASR systems are evaluated using WER (word error rate), and I was wondering if it makes sense to use a more fine grained approach. E.g. transforming both the ASR hypothesis and the ...
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1answer
828 views

Calculating Delta MFCC, correct formula?

I have seen different formulations of how to calculate delta MFCCs. In general, I want to be consistent in such a way that if I have other features to look at other than MFCCs, the way I would ...
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1answer
105 views

Mask a voice tone

What are the best ways to mask a voice tone, like what they do during news shows on television sometimes? Is there a way that is not reversibile? I think that a ...
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1answer
42 views

Matching two video conversations

I have two video where two person having conversation. They begin recording different times. I want to match these two videos' frame numbers, i.e. By matching audio. How can I do that?
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1answer
56 views

Bandwidth of time-varying harmonics of speech

For speech signals, do the harmonics have a constant bandwidth above some threshold (arbitrarily, -6 dB dropoffs)? Or does this bandwidth decrease with the upper harmonics? If this is not known, how ...
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2answers
216 views

Preprocessing signal before LPC

I have a signal that contains the speaking of three vowels. The signal is contaminated with dc offset and an interference at 60 Hz. I want to perform LPC for the synthesis of the spoken vowels. Do I ...
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2answers
328 views

pitch extraction for speech

I am building an LPC analysis tool, and am at the point of needing to do pitch period analysis. I was initially interested in using the Gold & Rabiner algorithm for this, and got a hold of a ...
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1answer
306 views

Speech discrimination from animal sound

i am looking for discrimination of speech from animal sound if you take a 150ms frame of speech and bird as shown below, what feature can i extract from them so that i can effectively state the ...
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1answer
2k views

Relationship between frame rate and sampling frequency in speech signal processing?

While I was going through the short time processing of speech signals, I got stuck at the relationship between the frame rate and the sampling frequency. Is there any relationship between them for ...
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1answer
154 views

Format location from LPC magnitude

I am confusing when I find the lpc roots magnitude, f1=0.9313, f2=0.9308, f3=0.91 and f4=0.95 but when I look at LPC envelope the f1 is higher than f4 in term of magnitude in dB. Could anybody explain ...
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1answer
107 views

How to compare two speech recognition softwares?

I am working on building speech recognition software for my company. In the meantime however, we will purchase some off-the-shelf commercial speech recognizers until the in-house one is ready. My ...
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0answers
8 views

Can Ivon's Justin text to speech voice be used by the Google's WaveNet Text to Speech API? [closed]

Is it possible to replace the voices available in Google's cloud TTS with the voice of Justin from Ivona ? If not direcly, is it possible in any other way to make the Justin voice use Google's TTS ...
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1answer
15 views

MATLAB & Power Spectral Density - Error - Getting the same range of values regardless of sound type?

I wrote code [attached at end of question] to take .wav files of child vocalizations and tried to see if there are valid ranges for the formants F1:F2:F3 for the 5 ...
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0answers
11 views

Analysis by synthesis in speech coding

I am trying to understand the analysis by synthesis in speech coding. In wikipedia it is mentioned for the CELP encoder: The main principle behind CELP is called Analysis-by-Synthesis (AbS) and ...
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0answers
38 views

Language detection from audio

I'm trying to do a simple test on language detection on a video input. Basically, given a .mp4 file, I need to process the audio tracks (can have primary and ...
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0answers
30 views

How can I quantify the distortion in a speech signal?

I am working with certain speech enhancement algorithms, for instance to remove stationary noise or impulse noise from a speech signal. Filters such as those can enhance the speech signal, but they ...
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0answers
64 views

Passband filter before Hilbert transform

I have a .wav file of human speech sampled at 48kHz. Before making the analytic signal with: hilbert(s_real) lets say in range 200 - 3,500 Hz, I probably need to do pass-band filtering, say 300 - 3,...
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1answer
123 views

Normalize speech features per dimensions or across dimensions

I am working on pipelines for audio processing (audio features that will go into a neural net to create some embeddings). We are using two features : Filter banks, or MFCC. I was wondering what was ...
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1answer
190 views

MFCC in Speech Recognition and ANN

I have a dataset with over 2000 examples of digits (0 to 9) recorded on wav files. I'm trying to process these files with MFCC so that later I could train an Artificial Neural Network (ANN). My ...