1) Equalizer matching involves finding the power spectrum of an example audio, then multiplying the input audio's magnitude spectrogram by a filter matching the example's power spectrum.
2) Noise matching involves denoising the input and example separately, then recombining their clean and noise components using the original signal-to-noise ratio.
3) Reverberation matching uses convolutive non-negative matrix factorization to decompose the input into a dry sound and reverb kernel, and convolve the estimated dry input with the example's reverb kernel.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Impr...a3labdsp
In the recent years, hybrid reverberation algorithms have been widely explored aiming to reproduce the acoustic behavior of real environment at low computational load. On this basis, exploiting the advantages introduced from hybrid reverberation structures, a novel approach for the reproduction of moving listener position through impulse responses (IR) interpolation has been presented in this paper. In particular, the presented methodology allows to remove
redundant information in large IR database also decreasing the memory usage and the computational complexity required to perform the auralization operation. The effectiveness of the proposed approach has been proved taking into account a real IR database and also providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
Hybrid Reverberation Algorithm: a Practical Approacha3labdsp
Reverberation is a well known eect that has an important role in our listening experience. Reverberation changes positively the perception of the sound, adding fullness and sense of space. Generally, two approaches are employed for articial reverberation: the desired signal can be obtained by convolving the input signal
with a measured impulse response (IR) or by synthetic techniques based on recursive lter structures. Taking into account the advantages of both approaches, a hybrid articial reverberation algorithm is presented aiming to reproduce the acoustic behaviour of real environment with a low computational load. More in detail, the early reflections are derived from a real impulse response, truncated considering the calculated mixing time, and the reverberation tail is obtained using an IIR lter network. The parameters dening this structure are automatically derived from the analyzed impulse response, using a minimization criteria based on Simultaneous Perturbation Stochastic Approximation (SPSA). The effectiveness of the proposed approach has been proved taking into account a real Italian Theatre impulse response providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Audio Morphing for Percussive Sound Generationa3labdsp
The aim of audio morphing algorithms is to combine two or more sounds to create a new sound with intermediate timbre and duration. During the last two decades several efforts have been made to improve morphing algorithms in order to obtain more realistic and perceptually relevant sounds. In this paper we present an automatic audio morphing technique applied to percussive musical instruments. Based on preprocessing of the sound references in frequency domain and linear interpolation in time domain, the presented approach allows one to generate high quality hybrid sounds at a low computational cost. Several results are reported in order to show the effectiveness of the proposed approach in terms of audio quality and acoustic perception of the generated hybrid sounds, taking into consideration different percussive samples. Mean opinion score and multidimensional scaling were used to compare the presented approach with existing state of the art techniques.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Impr...a3labdsp
In the recent years, hybrid reverberation algorithms have been widely explored aiming to reproduce the acoustic behavior of real environment at low computational load. On this basis, exploiting the advantages introduced from hybrid reverberation structures, a novel approach for the reproduction of moving listener position through impulse responses (IR) interpolation has been presented in this paper. In particular, the presented methodology allows to remove
redundant information in large IR database also decreasing the memory usage and the computational complexity required to perform the auralization operation. The effectiveness of the proposed approach has been proved taking into account a real IR database and also providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
Hybrid Reverberation Algorithm: a Practical Approacha3labdsp
Reverberation is a well known eect that has an important role in our listening experience. Reverberation changes positively the perception of the sound, adding fullness and sense of space. Generally, two approaches are employed for articial reverberation: the desired signal can be obtained by convolving the input signal
with a measured impulse response (IR) or by synthetic techniques based on recursive lter structures. Taking into account the advantages of both approaches, a hybrid articial reverberation algorithm is presented aiming to reproduce the acoustic behaviour of real environment with a low computational load. More in detail, the early reflections are derived from a real impulse response, truncated considering the calculated mixing time, and the reverberation tail is obtained using an IIR lter network. The parameters dening this structure are automatically derived from the analyzed impulse response, using a minimization criteria based on Simultaneous Perturbation Stochastic Approximation (SPSA). The effectiveness of the proposed approach has been proved taking into account a real Italian Theatre impulse response providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Audio Morphing for Percussive Sound Generationa3labdsp
The aim of audio morphing algorithms is to combine two or more sounds to create a new sound with intermediate timbre and duration. During the last two decades several efforts have been made to improve morphing algorithms in order to obtain more realistic and perceptually relevant sounds. In this paper we present an automatic audio morphing technique applied to percussive musical instruments. Based on preprocessing of the sound references in frequency domain and linear interpolation in time domain, the presented approach allows one to generate high quality hybrid sounds at a low computational cost. Several results are reported in order to show the effectiveness of the proposed approach in terms of audio quality and acoustic perception of the generated hybrid sounds, taking into consideration different percussive samples. Mean opinion score and multidimensional scaling were used to compare the presented approach with existing state of the art techniques.
The Short-Time Silence of Speech Signal as Signal-To-Noise Ratio EstimatorIJERA Editor
It is proposed in this paper to use a small portion of the audio speech signal to estimate Signal-to-Noise Ratio
(SNR). It is found that, the first 30 ms duration has enough information about the SNR in advance. The first 30
ms of a recorded speech usually comes from the silence rather than speech. This is because the speaker usually
starts the recording process or wait for it before he/she can deliver the utterance. For testing and comparing the
proposed estimator, different noisy corpora are built upon the TIMIT data. The average estimation of the
suggested algorithm proves to get better results as compared to the Waveform Amplitude Distribution Analysis
(WADA) and the National Institute of Standard and Technology (NIST) SNR estimators. The complexity of the
STS-SNR estimator is less than both as it only processes a small portion of the audio samples
echo types, how to cancel echo in each type, which is more complex, echo cancellation implementation in matlab
prepared by : OLA MASHAQI ,, SUHAD MALAYSHE
Cancellation of Noise from Speech Signal using Voice Activity Detection Metho...ijsrd.com
Speech Enhancement by suppressing uncorrelated acoustically added noise has been a challenging topic of research for many years. These are the primary choice for real time applications due to the simplicity and comparatively low computational load. This paper shows VAD (Voice activity detection) technique that can detect the non speech segment from the speech signal. It is also shown that it can work powerfully in an unpredictable noise ambience. The technique is mostly done in microprocessors or DSP processors because of their flexibility. But there are several advantages of FPGA over DSP processors like high cost per logic element related to these processors makes them improper for large scale use. From the experimental results, VAD method is implemented on the FPGA chip.
Digital signal processing through speech, hearing, and PythonMel Chua
Slides from PyCon 2013 tutorial reformatted for self-study. Code at https://github.com/mchua/pycon-sigproc, original description follows: Why do pianos sound different from guitars? How can we visualize how deafness affects a child's speech? These are signal processing questions, traditionally tackled only by upper-level engineering students with MATLAB and differential equations; we're going to do it with algebra and basic Python skills. Based on a signal processing class for audiology graduate students, taught by a deaf musician.
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
The Short-Time Silence of Speech Signal as Signal-To-Noise Ratio EstimatorIJERA Editor
It is proposed in this paper to use a small portion of the audio speech signal to estimate Signal-to-Noise Ratio
(SNR). It is found that, the first 30 ms duration has enough information about the SNR in advance. The first 30
ms of a recorded speech usually comes from the silence rather than speech. This is because the speaker usually
starts the recording process or wait for it before he/she can deliver the utterance. For testing and comparing the
proposed estimator, different noisy corpora are built upon the TIMIT data. The average estimation of the
suggested algorithm proves to get better results as compared to the Waveform Amplitude Distribution Analysis
(WADA) and the National Institute of Standard and Technology (NIST) SNR estimators. The complexity of the
STS-SNR estimator is less than both as it only processes a small portion of the audio samples
echo types, how to cancel echo in each type, which is more complex, echo cancellation implementation in matlab
prepared by : OLA MASHAQI ,, SUHAD MALAYSHE
Cancellation of Noise from Speech Signal using Voice Activity Detection Metho...ijsrd.com
Speech Enhancement by suppressing uncorrelated acoustically added noise has been a challenging topic of research for many years. These are the primary choice for real time applications due to the simplicity and comparatively low computational load. This paper shows VAD (Voice activity detection) technique that can detect the non speech segment from the speech signal. It is also shown that it can work powerfully in an unpredictable noise ambience. The technique is mostly done in microprocessors or DSP processors because of their flexibility. But there are several advantages of FPGA over DSP processors like high cost per logic element related to these processors makes them improper for large scale use. From the experimental results, VAD method is implemented on the FPGA chip.
Digital signal processing through speech, hearing, and PythonMel Chua
Slides from PyCon 2013 tutorial reformatted for self-study. Code at https://github.com/mchua/pycon-sigproc, original description follows: Why do pianos sound different from guitars? How can we visualize how deafness affects a child's speech? These are signal processing questions, traditionally tackled only by upper-level engineering students with MATLAB and differential equations; we're going to do it with algebra and basic Python skills. Based on a signal processing class for audiology graduate students, taught by a deaf musician.
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
Noise reduction is the process of removing noise from a signal. In this project, two audio files are given: (1) speech.au and (2) noisy_speech.au. The first file contains the original speech signal and the second one contains the noisy version of the first signal. The objective of this project is to reduce the noise from the noisy file
Echo and reverberation effects are used extensively in the music industry. Here we will design a digital filter that will create the echo and reverb effect on audio signals.
An Advanced Implementation of a Digital Artificial Reverberatora3labdsp
Reverberation is a well known effect particularly important for listening of recorded and live music. In this paper we propose a real implementation of an enhanced approach for digital artificial reverberator. Starting from a preliminary analysis of the mixing time, the selected impulse response is decomposed in the time domain considering the early and late reflections. Therefore, a short FIR is used to synthesize the first part of the impulse response, and a generalized recursive structure is used to synthesize the late reflections, exploiting a minimization criterion in the cepstral domain. Several results are reported taking into consideration different real impulse responses and comparing the results with those obtained with previous techniques in terms of computational complexity and reverberation quality.
Audio Noise Removal – The State of the Artijceronline
International Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
Audio Noise Removal – The State of the Artijceronline
International Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
This is my presentation on a Journal Club. It's based on the article: "Auditory inspired machine learning techniques can improve speech intelligibility and quality for hearing-impaired listeners". You can find all the references in the slide at the end of the article. I review very basic techniques in noise reduction, and how the techniques are implemented in the area of deep neural-network.
5. How? Signal Processing!
Input
Example
Trim
Resample
to 44.1
kHz
STFT Function ISTFT
Result
Normalize
R: hop size
: time frame
L: length of the signal
Smith, J.O. Spectral Audio Signal Processing,
http://ccrma.stanford.edu/~jos/sasp/, online book,
2011 edition
k: frequency index
w: window function
Preprocessing
5
11. Denoising
Spectral Subtraction
Noise profile estimate
Estimate clean power spectrum Noise suppression
factor
Fourier transform of the noisy
signal in one frame
In practice,
• Noise profile is estimated over multiple frequency bands.
• Spectral subtraction fails at low SNR regions by creating musical noises. This artifact is
reduced by post-filtering the spectral subtraction.
(Philipos C. Loizou, Speech Enhancement
Theory and Practice, 2013)
Additive stationary noise
( Esch and Vary, Efficient Musical Noise Suppression for
Speech Enhancement Systems, 2009)
11
13. Reverberation
Falkland Palace Bottle Dungeon
reverb sound
dry sound reverb kernel
(OpenAir database, www.openairlib.net)
Approximate in the
magnitude STFT domain
Convolution between
time frames of
magnitude X and H at
each frequency index
(R. Talmon, I. Cohen, and S. Gannot, “Relative
transfer function identification using convolutive
transfer function approximation,” IEEE Trans.
Audio, Speech, and Language Process, 2009.)
13
15. Reverberation Matching 1
Adry
Ra
Bdry
Rb
Dereverberation
Dereverberation
Ideal case – Perfect decomposition of reverb sounds into dry sounds and
reverb kernels.
Running out of letters!
input
example
Focus is on decomposing the magnitude spectrograms into magnitude spectrograms.
I took the signals back to time domain using the reverberated input phase information.
15
16. Convolutive Non-negative Matrix Factorization
Update Equations:
,
Paul O’Grady & Barak Pearkmutter, Convolutive NMF with a
Sparseness Constraint, MLSP Conference, 2006
Convolution of non-
negative matrices
Shift operator
Spectrum at time frame t
Matrix of size
Ly x k with all
its elements
set to 1.
16
17. Dereverberation
• Initialize with positive random values.
• Initialize with positive exponential decays.
• On each iteration, enforce anti-sparsity on ,
I dropped indices and absolute values, but they’re there.
17
18. Set of dry speech bases (trained offline)
Corresponding activation
Reverberated activation matrix
Dereverberation
We can do better by using more prior knowledge.
Convolution is associative
average R over multiple
frequency bands
(Paris Smaragdis, “Convolutive speech
bases and their application to supervised
speech separation,” in Speech And Audio
Processing. IEEE, 2007)
18
22. Summary
=>
Find power spectrums => Find EQ filter to match them. => Multiply the
EQ filter with every time frame in the input sound magnitude spectrogram.
=>
Denoise => EQ match the estimated clean and noise signals
individually. => Add the resulting input noise to the resulting clean signal
using their original SNR.
=>
Decompose to dry sound and reverb kernels => Convolve the
estimated dry input sound with the example sound’s estimated reverb
kernel.
22
26. Spectral Subtraction
noisy Signal clean Signal noise
A common assumption in most papers:
Noise and the clean signal are uncorrelated.
(Philipos C. Loizou, Speech Enhancement
Theory and Practice, 2013)
Fourier Transform over a segment of x(n).
AWGN. Same over all clean input segments.
Estimated Noise PSD.
In practice H is learned
over different
frequency bands.
26
27. Musical Noise Reduction
( Esch and Vary, Efficient Musical Noise Suppression for
Speech Enhancement Systems, 2009)
Aim: Retain the naturalness of the
remaining background noise.
How?
• 1
Detect low SNR frames based on the
noisy signal and the estimated clean signal.
• 2
Design a smoothing window based on 1.
Lower the SNR, longer the window.
• 3
Design a post-filter to smooth the low SNR
frames, i.e. an FIR low pass filter designed
based on 2.
• 3
Element-wise multiply the noise suppression
factor by 2.
Step 3
Enhanced Spectral Subtraction 27
28. SS + Musical Noise Reduction
G.*H Musical Suppression PostFilterSNR= 22 dB
Noisy Input
Much Better!
.^2 .^2
(
(
.^0.5
28
29. Metrics for Ideal Reverberation
time
Magnitude-dB
Energy Decay Relief
Energy Decay Curve
EDC at multiple frequency bands
29
30. Reverberation Model
• Time Domain Statistical Model
Where b(t) is a zero mean Gaussian noise. is related to reverberation time.
• Reverberation time = RT60= Length of time to drop below 60 dB below the original level.
Sabine Formula:
Volume of the enclosure
Effective absorbing area
Area
of each wall
Absorption
coefficient
Reflection Coefficients:
30
31. Image Source Method
Source
Microphone
Mirror image
of the original source
Actual path
Perceived path
Image source produces
another image source
(Allen, J and Berkley, D. 'Image Method
for efficiently simulating small‐room acoustics'. The Journal of the
Acoustical Society of America, Vol 65, No.4, pp. 943‐950, 1978)
(Pictures from: Alex Tu, Reverberation
simulation from impulse response using
the Image Source Method)
Parameters that control which image source in which dimension
Reflection coefficients of the six surfaces in a rectangular
Time delay of the considered image source
31
32. Non-Negative Matrix Factorization
,
• Applying Gradient Descent under positive initial conditions for W and H and a ‘clever’ learning rate results in
the following multiplicative update rules,
(Lee and Seung, 1999)
Normalize W
32
33. Why NMF? (Lee and Seung, 1999)
Visually meaningful.
Decomposition can only be
positive. Part based
presentation.
Statistically meaningful.
Eigen faces are in the
direction of the largest
variance. Subtraction can
occur.
33
39. Spectral Subtraction
SNR= 22 dB
Musical Noise –
mainly at low SNR regions
Noisy Input
Denoised-ish?
Go back to time domain
Use noisy input phase
H – Noise Suppression Factor
.^2 .^2
(
(.^0.5
39
40. With Musical Noise
SNR= 22 dB
Same results, better colormap?
Without Musical Noise
Noisy Signal
40
Editor's Notes
Why? What?
Fix the STFT equations
Use the powerful yet so simple equalizer matching to do denoising as well.
Well, now we can’t ignore time here anymore. Reverbs are usually longer than a time-frame and are presented in a convolutive manner. FIR filtering here gives you too many taps, and even when inversing you have to deal with whether its minimum phase and invertible and …
Use a sound that gives you a less artifacty result.
Def of conv2
Put some sounds here…
If you’re interested I designed a user inteface to play with.
Might want to get rid of details and only show some to intrigue questions
is the total amount of signal energy remaining in the reverberator impulse response at time
(Smith, J.O. "Delay Lines", in Physical Audio Signal Processing, http://ccrma.stanford.edu/~jos/pasp/Delay_Lines.html, online book, 2010 edition)