Questions tagged [speech]
The speech tag has no usage guidance.
-1
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0answers
10 views
HMM Model HTK TOOL KIT [on hold]
I have trained the hmm model on audio files.
I followed the following tutorial for training.
Link
So the problem is that I am getting the negative accuracy.
...
0
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0answers
19 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 ...
0
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0answers
24 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 ...
0
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2answers
34 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 ...
0
votes
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 ...
0
votes
1answer
63 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
vote
1answer
162 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-...
1
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1answer
48 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 ...
0
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0answers
44 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,...
2
votes
3answers
196 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_\...
0
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0answers
39 views
audio filter designing help
buzzer frequency range
recording of audio signals(environmental sounds with speech) with buzzer soundI have audio signals(environmental sounds with speech) with buzzer sound, I have to attenuate only ...
0
votes
1answer
545 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:
...
0
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0answers
34 views
Calculate RMSE and DTW before or after signal normalization
I want to calculate RMSE and DTW of clear speech signal and noisy speech signal. Im normalizing signal using division by max possible value (for 16bit audio its 32756) to have signal in range [-1;1]. ...
-4
votes
1answer
72 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 ...
0
votes
1answer
101 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 ...
0
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1answer
882 views
0
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0answers
65 views
Speech quality evaluation of audio of small size
I want to evaluation speech quality of audio of very small size.
Code for PESQ and STOI available on net need atleast 4000 samples to evaluate.
Actually, I want to split audio of large size into ...
0
votes
0answers
47 views
Initial average Source Distortion Ratio for speech mixture
I am reading a paper where it says, that the initial Source Distortion Ratio (SDR) averaged over the mixtures of speech is x dB. I read the original paper by Vincent where he defined SDR. I thought to ...
3
votes
1answer
57 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
97 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 ...
3
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1answer
119 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 ...
-1
votes
1answer
87 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
votes
1answer
161 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
votes
1answer
526 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 ...
1
vote
1answer
233 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
votes
1answer
105 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
vote
1answer
51 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
vote
1answer
850 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
662 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
votes
1answer
220 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 ...
-1
votes
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 ...
0
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0answers
73 views
Low power/processing methods for speech-music discrimination on Raspberry Pi
I am to implement an algorithm that controls a vacuum cleaner robot to move towards voice, ignoring other sources of audio including noise and music. I have worked on the audio localization - however, ...
1
vote
2answers
752 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/...
1
vote
1answer
605 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 $x(n)$ ...
2
votes
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
58 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
votes
1answer
177 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 ...
1
vote
2answers
440 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.
0
votes
2answers
1k 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 ...
4
votes
0answers
330 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
votes
2answers
281 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
votes
3answers
840 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 ...
5
votes
1answer
350 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 ...
0
votes
1answer
754 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
votes
1answer
156 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
votes
1answer
102 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
votes
1answer
328 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 ...
0
votes
3answers
1k 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
vote
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. ...
1
vote
1answer
93 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 ...
