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Questions tagged [speech]

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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 ...
0
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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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1answer
169 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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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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1answer
241 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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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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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
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 ...
1
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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
50 views

About speech preprocessing in mobile phones

I understand that cell phones transmit speech signals in digital form through 2G/3G or other standards. But I could not find anything about the signal processing method lying in between the analog ...
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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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3answers
215 views

Phase change rate of human speech

I have a .wav file of human speech, sampled at 48kHz, with useful signal up to 5kHz, and almost no noise. If I do Hilbert transform in MatLab, I get the Analytic signal: $$ s_\mathrm{a}[n] = s_\...
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1answer
948 views

How to convert NIST to RIFF wav file

I have a dataset of audios(namely TIDIGITS, collected at Texas Instruments, Inc.) and they are in .wav format. However, when I try to read them from python, I get the following error: ...
-3
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1answer
87 views

How do i remove the DC component from an audio signal using Matlab2016?

I have been doing speech processing in matlab2016, and am stack on how to remove DC component. I tried using the the function y=x-x(mean) but it seems not to ...
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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
1k views
2
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1answer
61 views

How exactly does sound get encoded by a cochlear implant processor?

I've been wanting to know more about how sound is processed by a cochlear implant processor, how they are programmed, and how they stimulate the magnets and electrodes inside. How do you take pure ...
2
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0answers
112 views

Speaker diarization tool for acoustic analysis

I'm researching certain acoustic prosodic features during live clinical interactions. Because of the setting, the recordings always have cross channel bleed, and in some cases everything is recorded ...
2
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1answer
151 views

Speech signal formant spectrogram. How can we determine F1?

The formant spectrogram in this page points F1 and F2 as below; In bottom-right F1 movement, I thought F1 could be below blue one. But how can we determine F1 is upper one not below one? Is it ...
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1answer
115 views

symmetrical wav audio in time domain and Phase spectrum significance

I have plotted many wav files in MALTAB and tried to understand their time domain plot , Magnitude spectrum and Phase spectrum. But I don't get why time domain plots of speech wav files are ...
0
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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 ...
4
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1answer
723 views

Speech recognition: words vs. phonemes

One could train a speech recognizer with words or phonemes. I guess that they both have their pros and cons: with words there would be a lower error rate as phonemes can be put together building a ...
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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 ...
0
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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 - ...
1
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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 ...
1
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1answer
958 views

The minimal signal to noise ratio (SNR) for people to understand a speech in the noisy background

This question seems easy to find an answer but I got lost in the end. I am trying to find the threshold for people to understand speech in a noisy environment. For example, assume the background noise ...
2
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0answers
756 views

Speex echo cancellation takes a long time (~20s) to converge

I have been experimenting with Speex's echo canceller during the past few days. I got the echo canceller to work, however, it takes a long time until it actually begins filtering out the echo. It ...
0
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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
2k views

Compare two audio tracks

I need to compare two audio tracks and compute the difference. The difference is how close they are in terms of pronunciation, rhythm etc. ignoring speaker voice characteristics. The first track is ...
2
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2answers
857 views

Understanding LPC for Formant Estimation

I went through the Matlab tutorial on Formant Estimation using LPC Coefficients. Though I vaguely understand the details, it's not entirely clear why we need to do this. From http://person2.sol.lu.se/...
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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 $x(n)$ ...
3
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0answers
3k views

What's spectral “tilt”?

I am looking at article Speech-in-noise intelligibility improvement based on spectral shaping and dynamic range compression. In paragraph 2.2 the article mentions "tilt" of the spectral envelope. The ...
0
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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 ...
0
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1answer
236 views

audio quality evaluation

I am creating a dataset, where in many speakers are recording their speech signals using different microphones and in different environment. I would like to evaluate the quality of audio and discard ...
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2answers
538 views

Why discrete cosine transform may not maintain locality

In paper Convolutional neural network for speech recognition, they say I didn't understand the highlighted sentence.
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2answers
2k views

FFT spectrogram in log frequency space. How?

I was reading a paper and the authors say that they extract FFT spectrograms and log-scale in frequency domain. I understand how the general spectrogram is computed, but how to make it log-scale in ...
3
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0answers
352 views

Implementing blind deconvolution in MATLAB

I want to implement blind deconvolution for the signal $ r(n) = h(n) \star s(n) + a(n)$ in MATLAB where $r(n)$ is the recorded speech $h(n)$ is impulse response of room acoustics $s(n)$ is desired ...
4
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2answers
309 views

Matching LPC frequency response to raw sample data frequency response

Background: The intention is feature/formant extraction from speech input. I have used an FFT on raw samples in the past and have done simple peak picking from a weighted average of frequencies over ...
2
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3answers
1k views

Speech corpora for VAD

I'm doing a student research project connected with VAD. And I want to find speech corpora with added noise. I found many references to the NIST Speaker Recognition Corpora. But unfortunately, all ...
4
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1answer
385 views

What does MATLAB display when plotting a sound signal?

When you load the speech file 'mtlb' in MATLAB (which is by the way the voice of a woman saying the word 'MATLAB'), and plot the data, you get the following plot: My question is this: what does 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 ...
2
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1answer
162 views

Using 8 bit samples for speech comparison

I am making a speech comparison app in javascript. I found this dsp.SE post: Speech comparison algorithm for rating on similarities, where I read it is best to use mfcc and dtw algorithms. I ...
0
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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 ...
0
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1answer
356 views

Obtaining signal energy modulation for voice activity detection

Im currently working on VAD (voice activity detection) topic. Since I'm beginner in digital signal processing, I've started to wonder if calculating single frame energy (after aplying FFT to it) and ...
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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 ...
1
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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 ...
0
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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 ...