Bioacoustics - Science topic

Hello Jorian; Ahmad's response is thorough and is sensitive to the complexity of your question.

1. How large is the home range of your species of interest? Do the sexes differ regarding this trait?

2. How widespread is the species' preferred habitat within the study area?

3. How expensive will it be to install one more sampling site?

Questions like these will influence your decision.

Best regards, Jim Des Lauriers

Hello dear friend

what country and present your species sample

There were two related talks that presented very promising results at the 2022 DCLDE workshop (see https://www.soest.hawaii.edu/ore/dclde/program/ )

Léa Bouffaut- Listening at the speed of light: baleen whale monitoring using distributed acoustic sensing

William Wilcock- A Community Test of Distributed Acoustic Sensing on the Ocean Observatories Initiative Regional Cabled Array Offshore Central Oregon

Differences in distress: Variance and production of American Crocodile ( Crocodylus acutus ) distress calls in Belize

Visit to that article

Mariano J. Feldman, when considering the effects of background noise on detection capabilities of specific sounds, you would want to measure sound levels over the frequency range of the signal (sound) of interest. For example if frogs produce sounds from 50 Hz to 1 kHz, you may find it useful to look at background noise over that bandwidth. Filters can be used to measure the RMS levels over specific bands of frequencies, but PSD values are probably more useful as you can integrate those values over the desired frequencies. Also 1/3 Octave and Octave levels are useful for monitoring patterns in sound levels over standardized frequency ranges. I am not sure if you are familiar with R or Matlab, but I would search for a program called PamGuide by Merchant et al. if you do not want to write your own code. There is also a paper associated with that software that is useful for understanding those measurements, also by Merchant et al. Lastly, when considering background noise, you will need to be aware that the signals of interest will likely influence those background noise levels. Picking a time when those signals do not occur will give you an understanding of baseline sound levels in the absence of those sounds.

Dear Peter,

Thank you so very much for your kind answer. I have already ordered plant spikerbox from the backyard brains. But the plants I am primarily studying are Tamarind plants (Tamarindus indicus). I am a bit worried if the plant spikerbox can pick up the low frequency signals of this plant. My Tamarind plants have shown some interesting behaviour like keeping leaves open of cold stormy nights and closing the leaves for few days even if there were ample sunlight. This behavior have been repeated for last two months. So, if I may ask this suppl8mentary question that if the spikerbox cant pick up tamarind plant signal, is there any way to tweak the device that it can pick up tamarind electric signals?

Dear Hans,

many thanks. My e-mail is pavel.jaska@nature.cz. I am looking for your articles. Now I have not so much information about all this cannon methodology, efficiency etc. I will dive deep in this all and then I will make some summary about that. I will send it to you when it will be prepared.

many thakns for all

Pavel

Hi Vineeth,

In the paper

Narins PM (1995) Temperature dependence of auditory function in the frog. In: Advances in Hearing Research (GA Manley, GM Klump, C Köppl, H Fastl, H Oeckinghaus eds.) World Scientific Publishers, Singapore 198-206, the authors present examples of how temperature afects frog call parameters. The following is one example from this paper of how to calculate the effect of a 10 degree C temperature change (the thermal Q10) for a frequency shift of s octaves:

Avisoft SasLab Pro 5.2 software has got inbuilt lowpass, highpass, notchpass and bandpass filters critical in elimination of unwanted noise. Open the Avisoft SasLab Pro 5.2 software, the choose the edit menu. Under edit menu select Filter. Under filter, choose the Time domain IIR or FIR Filter. If its the background noise, then you may record the room tone and filter it out. Similarly the Avisoft UltraSound Gate allows you to attenuate unwanted signals.

Hi Jacques,

In fact, we did just these kind of playback experiments as part of a project in which we demonstrated that the Concave-eared torrent frog, Odorrana tormota, from China produces ultrasonic call components and behaviorally responds to the US components alone. See:

Feng AS, Narins PM, Xu C-H, Lin W-Y, Yu Z-L, Qiu Q, Xu Z-M and Shen J-X (2006) Ultrasonic

communication in frogs. Nature 440: 333-336.

Hope this helps-

Peter

Hi Anish,

It is interesting that vocalization frequency tends to get lower as one proceeds up an altitudinal gradient. Moreover, dominant frequency also decreases with increasing body size. For many animals, body size also increases as one proceeds up an altitudinal gradient, due to harsher conditions at higher altitudes. Harsher conditions favor larger animals with smaller surface area-to-volume ratios.

In a study (refernce below) we did on hearing in tropical treefrogs found along an altitudinal gradient in Puerto Rico, we concluded:

"We suggest that the animal’s body size determines the frequency particulars of the call apparatus and the inner ear."

I hope this helps,

Peter

Hi Thomas Luypaert , The window size in a FFT analysis represents a number of samples, and a duration. To set the size you would need to look at the fundamental frequency in your system, its intensity and any changes it may experience in time. The window size expressed in samples along with your sampling rate (samples/second) would give you information to calculate the window duration.

Also, from the spectraplus page: " The frequency resolution of each spectral line is equal to the Sampling Rate divided by the FFT size. For instance, if the FFT size is 1024 and the Sampling Rate is 8192, the resolution of each spectral line will be: 8192 / 1024 = 8 Hz. Larger FFT sizes provide higher spectral resolution but take longer to compute."

Hope this helps!

Abel has it! :-) Jim Des Lauriers

Hi Valentijn,

Ben zelf geïnteresseerd om geluidsopnames te maken van knoflookpad in Vlaanderen.

Kan je me mailen op lily.gora@vlaanderen.be, aub?

By asd device you will get .asd file and you can convert that ascii asd file to txt file with help of viewspcpro software. Then import that data into excel sheet.then you can calculate any SVI.

Russell Gray I think she is asking more if machine learning is feasible? My immediate answer is yes. We work on multi-species assemblages, we might have as much as 6-7 species calling within just a few seconds, it challenges the algorithms a a bit. Especially as the data is from omni-directional microphones so there's a lot of background noise. Not impossible though, just need to ensure the training is done thoroughly. Have a look at the papers by Bedoya 2014 (Automatic Recognition of Anuran Species based on Syllable identification). We are also working on implementing deep learning on our data - they've done it for bats in our lab (I think the paper is in review) and there was just a paper released on using it on birds I think? (I can't find it or remember the author off the top of my head, might have been in Methods in Ecology). Birds are in my experience a pretty good model species though as their calls are quite complex so easy for the algorithms to identify, we have more issues with the frogs as the call itself is often quite "simple" in it's pattern (then put together into a more complex call series), I don't know how primate calls look like in spectrograms though.

Hello the right sounds can produce tremendous improvements in growth, and the wrong sounds can do just the opposite. Plants are more aware of their surroundings than we think, probably much more so than us!

There is a highly instructive user manual on the site. The reference for the initial publication is:

Tchernichovski, O., Nottebohm, F., Ho, C. E., Pesaran, B., & Mitra, P. P. (2000). A procedure for an automated measurement of song similarity. Animal behaviour, 59(6), 1167-1176.

I have studied with Olga Fehér who was herself a PhD student of the first author on the paper. I have learned a few things the hard way. First off, the initial setting of this software is designed for the analysis of zebra finch song. If you are working with a different species you may need to adjust them. Also, the detection and classification algorithms are good, but by no means perfect. Therefore I strongly advise that you keep the results by eye!

Greetings from the UK!

Tarandeep

For most analyses, the biggest problem with clipped recordings is not the loss of reliable information about relative amplitude ("dynamics"); it's the corruption of the spectral content of the recording. Clipping introduces spurious energy into the recording at frequencies that are harmonics of the true signal. In some cases, sounds that are purely tonal (no harmonic content) will end up having strong harmonics in the clipped recording. Sounds that truly do have harmonics (like many parrot vocalizations) will be recorded with distorted relative powers in different harmonics. Most kinds of analysis that aim to classify sounds or measure their relative similarity (e.g., spectrogram cross-correlation, dynamic time warping, etc.) may be severely affected by such distortions.

What about "fixing" a clipped recording? Many digital audio editing programs (e.g., Audacity, Audition) include tools for "fixing" clipping. However, these tools are not intended to provide a perfect reconstruction of what the unclipped signal would have looked like. They're meant for rendering the harmonic distortions of brief periods of clipping in music or speech recordings inaudible to the human ear. This is very different from what would be needed for a quantitative measurement and classification analysis.

Although there may be some kinds of analysis that could tolerate clipping (e.g., analyses focused on temporal patterning of sounds that don't measure spectral content), in general I recommend against using clipped recordings for pattern analysis and classification. You're likely to get quite different results from what you would obtain using unclipped recordings of the same signals.

I think that Polajnar and Maynard-Smith have it about right. I would say simply that communication occurs when the behavior of one organism produces a stimulus that changes the behavior of another organism. That includes both behavior due to phylogeny and due to ontogeny. In particular, it includes verbal behavior. Disagreeing with Butler, I also would argue that an organism may communicate inadvertently. Think of Clever Hans the horse, for example. It's owner communicated inadvertently.

Hey Cristian,

I've found that devices marketed specifically for playing music are often manufactured with a "curved" frequency response. Some set of frequencies within the range you provided will likely be reproduced at a lower volume than others.

The device with the "flattest" frequency response I've used is a predator caller. They range in quality along with price, but depending on your needs I'm sure there's something that will work for you. The brand I'm familiar with is FoxPro, specifically the Inferno model, which is a single horn type speaker. Worked great for producing reliable sounds between 1-7 kHz at 100dB. Have a look at some of their user manuals for more specifications.

Hope to help!

-Andrew

At present we are working with MonitoR in R. It looks promising for a basic sound pattern search across multiple files of unattended recordings.

The only one interactive software we found is X-Bat, a toolbox developed for Matlab but not maintained anymore. Any hint for finding something similar ?

Hi Adwait!

Have you checked warbleR ? This is a free R code developed by Marcelo Araya-Salas to streamline analysis of animal acoustic signals. 

Probably this is what you are looking for, although you must have some knowledge of R to use.

All the best!

Muchas gracias Gian, you work is very interesting!

Update; I buy an used Olympus LS-100, after reading helpful reviews in birds pages and bioacoustics forums in general. And my microphone is a shotgun Sennheiser (the pic is attached to this reply, i don't know the model). Thanks to all!

Thank you all for the excellent feedback and information - I will attempt to implement the suggestions in the coming months and be in touch if further question arise. 

Hi Laura,

I agree with Pavel regarding the channels and amplitude. I also don't use SoundRuler (sorry!) but I thought it might be useful to add that you need to be sure there is no background noise overlapping your calls of interest. If for example these recordings were made at a zoo, there may be visitors chatting, or in the field there could be other animals calling etc. on your recordings. If there is, then you can either need to filter it (if it does not overlap in frequency with the monkey calls), or if that's not possible, you could manually extract the frequency measures, or simply eliminate those calls from your analyses. It might be that you can use the read-out on all your "clean" calls (i.e. no background noise, one individual calling at a time) and in that case it could be just as simple as you say!

Good luck with your interesting project!

All the best,

Esther

Are you looking for the propagation transfer function between the acoustic source and the acoustic receiver?

That transfer function is often called the "propagation loss" or "transmission loss". It depends on the the kind of acoustic source you are dealing with, such as a point source or a plane-wave source for instance. And it depends on the medium through which the sound is propagating, such as through homogeneous air or other fluid like water, or perhaps through layered media in earth sciences (the ocean in oceanography).  And it generally depends on the frequency as well, as you say.

If it is the propagation transfer function that you want, then you must describe your propagation scenario in more detail, including a description of the source, the propagation medium, and source-receiver separation distances that are of interest.

Ronald

Articles on ear evolution:

Regards,

Joachim

HI, you might want to check with Phil Lobel and David Mann, they would be the most likely to have recorded the species.

Thank you everyone, it really helps! I will soon let you know my choice and how does it sound.

Please have a look at these links.

Dear fellows,

I've been watching your discussion related to "Do fish measure reactive intensity?" and found it interesting when talking about "total sound intensity from a pure acoustic point of view". When it comes to sound reception/sensing in fish you've forgot the "most" important sensor beside the swim bladder and the lateral line, i.e. the otoliths, which act like an accelerometer. There is some literature related to this which may clarify/elucidate sound reception in fish in relation to total sound intensity. I'll like to see you++ continue with this discussion.

Squirrelfish are a classic group for study of sound production. My old major prof Howard Winn and students did some of the classic work, and it has recently been taken up by the Parmentier establishment. Be careful about using common names when asking an international audience about a fish. I can come up with at least 10 names of the oyster toadfish.

Some of the Wilcoxon sensors were developed with the US Gov't, and have ITAR restrictions associated with them.  They are not available internationally.  Applied Physical Sciences also has some sensors, but they too have ITAR restrictions.  The Microflown/Hydroflown sensors (from the Netherlands) have never been proven to be effective in underwater applications. 

My recommendation would be to get several good pressure sensors and build your own vector sensor based on pressure gradient measurements.  (I believe that's really what the PAS is.)

sure, if you send me a template file I will take a crack at it and send you a set of m-files. I would consider it a fair trade, I haven't had a chance to work with whale song much lately.

Hello,

You could use the R packages seewave, tuneR, soundecology and ineq (https://www.r-project.org/). “R” is free and useful in many ways. Plus, there are numerus articles using these R packages for bioacoustic and ecoacoustic research.   

Nevertheless, for a more basic and user friendly approach, you could use Audacity (http://www.audacityteam.org/).    

Best regards,

Aggelos

Here is my dads' paper actually, turns out some of these end up archived in strange places:

thanks professor!

Great question!  I have many good and bad stories about various devices on small animals, but have come to the conclusion that the traditional methods are not as fool-proof and too expensive compared to what is new and better.  And I have a (hopefully great) suggestion: use passive harmonic dipole.  These are the same things that avalanche jackets have in them and you would need to buy the whole package (including the transceiver), but the costs might be cheaper than traditional radio tracking depending on how many animals you want to track.  Check out the Herp Review article (Herpetological Review, 2011, 42(4), 522–525) by GOURRET  and others from Australia (attached) and/or just Google "animal tracking passive harmonic dipole" and see what you get.  These devices are smaller than a grain of rice and can be placed on your study animal in a variety of ways, although a thread belt might be best for your sallies.  This is so the diode will eventually just fall off and you can find it again - and re-use on another animal.  I have a few colleagues who are now swearing by these and can vouch for their effectiveness for small amphibians.  Plus, you won't need to surgically implant them - a huge bonus for you and the animals!  Please let me know if you have any questions and I can help you with contacts that might be able to give you the best vendors etc.  Also, sometimes you can buy old and used avalanche finders from ski resorts and use those for much much cheaper as well.  Hope this helps.

Ok, that's one I had not seen. I was going in with a bent based on some of Dr. Bass' work where he tested various modulations of the same tone on midshipmen, and basically the pure tone got 100% of the gravid females to come hither. the more modulation was introduced, the less interested the females got. at this point I'm wondering if a modulation effect isn't present that fades with maturity. I'll send you some stuff under separate cover.

Hello,

Bioacoustics is a field that deals primarily with animal vocalization, sometimes in relation to anthrophony. For example, one could conduct bird song recordings to assess the masking effect of noise in animal communication.   

Some researchers conduct their recordings using remote sensing techniques. They deploy several omni-directional sensors in multiple sampling spots across the landscape in order to simultaneously conduct recordings of all significant events taking place in the soundscape, for a long period of time.

In order to obtain a realistic portrayal of the overall soundscape, meaning to capture the majority of the acoustic events taking place in a 24-hour period, it is necessary to organize your sampling protocol accordingly.  

The duration of the analyzing procedure depends on the software you are using. “R statistics” (with the appropriate packages) is very good and free but occasionally time consuming.       

For further information on recordings conducted for Bioacoustics, Soundscape Ecology and Ecoacoustics research, refer to articles written by Bernie Krause, Almo Farina and Henrik Brumm.

Thanks very much for your help.

as mentioned before there are two problems, storage and batteries. Among small digital recorders the best one is the SONY PCM-M10, unfortunately now discontinued but still available in US. Its power consumption is among the lowest: 60-70mAh, that means >30 days continuous recording with 6 or 8 D size batteries (max 18Ah at 1.5V), that brings weight and size high. As an option you could use Lithium D size batteries, expensive but with double power than alkaline (up to 19Ah at 3.6V).

As for the storage the M10 is one of the very few that is declared max 32GB but can record on a Sandisk Ultra 128GB microSDXC formatted by a PC in FAT32 mode. The max microSD size now available is 200GB but I never tested it in the SONY. Huge size available; however, to record 30 days you necessarily need to switch to MP3-320K with a bandwidth of 15 kHz, that is more than with 22k PCM sampling.

In 22kHz PCM stereo (few recorders can record in mono !) you need 228GB/month that would be possible with a standard 256GB SDXC card (these are also available up to 512GB now). But recorders are larger and more power hungry.

Solutions based on microcomputers need too much power. In any case the first feature you must check is power consumption. A recorder that would partially fit your needs is the Tascam DR22WL, it records in mono and thus 1 month could fit in a 128GB microSDXC card. Unfortunately it requires near to two times the power required by the SONY M10. If you solve the battery problem the WL 22 is the good solution.

Another option to be deeply tested before going on is to use the ZOOM H1. Smaller and lighter than the SONY, It is declared to max 32GB microSD however it can record on larger memories as the SONY (to be tested on specific memory brands and to be tested for power requirements).

Eventually write me privately for alternative options.

Gianni

It is not necessary that sender and receiver both benefit from a new communication process. Many signals arise by 'exploiting' the sensory system of the receiver. For example, guppy males attract more attention by females when their color matches that of the most preferred food of the females. This way, males can evolve a certain color that later plays an important role in sexual selection. The color trait (it depends on your definition from which point onward you call it a 'signal') may spread even if mating with colored males would be disadvantageous for females. A recent review article on this 'sensory exploitation hypothesis' is Ter Hofstede et al. (2015) in Current Biology.

 

Such a  pattern can  result  from  the  error  inside  the  transducer-amplifier channel. The most  probable  the  transducer  is damaged or  not  connected correctly. The patterns are reflected the noise or  electrical disturbaces.

Dear Karyn,

Our foundation has been funding for several years a project which has developed an automatic detection system. This has been designed to detect, extract and store in a database killer whale vocalizations but theoretically it should work with any sound event. It would be great to have one of your recordings to make a test and, if if works, we can find the way to process your  recordings, as the system stills under development and there is no a version for distribution yet.

J

As suggested by Israel, if the "nature" of your noise does not change (assuming frequency remains the same as the signal but not the amplitude or phase) then a simple amplitude based filter should work. You may also try phase-based filters.

If the noise (or even if the signal) changes its characteristics then go for statistical filters which can even be adaptive! The first one to try out is Wiener filter mechanism with adaptive filter algorithms like LMS, NLMS and RLS. These are the best ones (my personal experience). See these for more information 

and

and

Thanks all.

Moving forward we will be using a dual metrics criteria that includes the interim NMFS sound thresholds as well as Southall et al. 2007 values (SPLrms and SPLpeak).

We'll be keeping an eye on the draft NOAA guidance as well to see what the final version outlines. For those interested this is the draft guidance: http://www.nmfs.noaa.gov/pr/acoustics/guidelines.htm

Hi Russ Charif  Thank you very much for your suggestion and information to me .

i will be message you in research gate , about my problem using Raven Pro.

Regards 

Muhammad

Hi Robert, here are some references. Please check and let me know of your idea.

the first step is to define what you want to demonstrate/show: their behavioral function/meaning ? their stability/stereotypy for species recognition ? their specificity versus other similar species ? their possible use for automatic species recognition ?

A basic spectrographic analysis might be enough for most of these cases, but you need a suitable sample size. You need N individuals (or N recordings of different individuals, maybe with some degree of individual replication).

According to the shapes you see on the spectrogram, and also according to your listening of the sounds, you then need to identify some measuring points that enable you to characterize the vocalizations and analyze them statistically.

that's the basic strategy, then you need to tune it according to your research goals.

Gianni

Hi Steven,

Thanks Abhishek Raj for tagging my paper.

In our lab in Australia we use call playback for a wide range of Litoria species. However, there are a lot of species here that don't respond to call playback.

For our auditory surveys we estimate the number of callers prior to playback for three minutes, play calls (or imitate calls) for one minute, then record the number of answering males. We have found this to be useful as an abundance measure for calling activity, but it is always coupled with visual encounters. Our females and juveniles dont make sound so VES is needed and our males are very hard to spot in VES as they startle easily and hide (but females and juveniles don't hide). This covers us for male presence as long as we statistically account for changes in weather and change in calling as e.g. dropping temperatures reduce the amount of calling (however this cant account for changes in male energetic constraints).

There are a couple of things to consider if using call playback, I'm not sure if your interested in this side of things.

1. The number of calling males you use in playback (as previously mentioned)

2. The type of call you use (advertisement call, aggressive call, release call, unenthusiastic call) If you don't know what the calls mean, maybe choose the most common calls in good weather/breeding periods? A pilot study could help determine this and even make a neat note for call types.

3. Not all males within a chorus area actually call (satellite male behaviour)

4. We are going to modify our methods to also estimate the number of males calling during call playback. Sometimes males call during call playback but not afterwards, and kind of 'mirror' our calls, so actual response can be lost if only recording afterwards.

5. When I take volunteers out, I never rely on their ears. I have found with my species that untrained people massively overestimate the number of calling males as their calls are very loud and you feel your surrounded by 20, but its really 8

Good luck with your surveys.

If there is only one animal calling there, it is possible that you have a kind of nonlinear phenomena common in mammalian vocalizations (biphonation). So, you probably have two independent fundamental frequencies (as you can see in the vocal units from the middle of the call on). It seems that at the begining of the sequence the animal is producing tonal calls that the fundamental is the F0. Then you have the biphonation happening.

Please, find more about biphonation in these papers:

Riede, T., Herzel, H., Mehwald, D., Seidner, W., Trumler, E., Tembrock, G. & Bo¨hme, G. 2000: Nonlinear phenomena and their anatomical basis in the natural howling of a female dog-wolf breed. J. Acoust. Soc. Am. 108, 1435—1442

Fitch, W. T., Neubauer, J. & Herzel, H. 2002: Calls out of chaos: the adaptive significance of nonlinear phenomena in mammalian vocal production. Anim. Behav. 63, 407—418.

It sounds like an interesting project, however it will be much easier for people to help if you tell us: what frequency and duration of sampling will you need, in order to address your research questions?

In favourable conditions you can get good video and sound recording with almost any compact digital camera, even with older models as Tessa mentioned. And you can easily mount a compact camera on a low tripod or other stand and add a rain-proof cover (or choose a waterproof model), place it near your study animals, turn on video mode, walk away and leave the camera to run continuously...

...until the battery runs down or memory card fills up... maybe half an hour, maybe an hour or two, depending on camera, memory and video settings.

Much more challenging if you want to record continuously all day, day after day. You will need external power and high capacity data storage... and afterwards you will need to spend many days watching/listening in order to extract the data of interest.

Some other considerations: can you work near these animals (set up equipment, change batteries, etc) without disturbing the behaviour you are studying? Can you predict fairly accurately when and where the behaviour of interest will occur?  How near and how far from the recording equipment do you expect the animals be heard/seen? Do you want the vision to be clear and close enough to identify age/sex/other characteristics? How much ambient noise from wind or road traffic or other sources? How many animals do you want to record concurrently?

Muchas gracias Juan Josè

In Yellowhammer Dialects project, I am exploiting many sources of recordings:) Surprisingly, some yellowhammer song are also on Flicker, however I bet this is not the case of African warblers.

One is on AvoCet (http://avocet.zoology.msu.edu/recordings?scientific_name_search=Acrocephalus+baeticatus&name_search=&Search=Search+Recordings)

The best would be to contact British library - they have about 47 songs: http://cadensa.bl.uk/uhtbin/cgisirsi/?ps=jvO7LMNQhB/WORKS-FILE/325200049/123 I will send you a contact to a responsible person through email, Nika:-)

Dear Nahid, they're manca stages of terrestrial isopods.

This is for A. cisternasii but you may find similar papers for A. obstetricans:

Female Courtship Call of the Iberian Midwife Toad (Alytes cisternasii)

Jaime Bosch and Rafael Márquez

Journal of Herpetology

Vol. 35, No. 4 (Dec., 2001), pp. 647-652

You may use Shannon entropy measurements to calculate note disorder within the song. Please take o look in:

SILVA, M. L. ; VIELLIARD, Jacques Marie Edme ; PIQUEIRA, J. R. C. 2000. Using shannon entropy on measuring the individual variability in the Rufous-bellied Thrush Turdus rufiventris vocal communication. Journal of Theoretical Biology, 207(1): 57-64, 

 While much of the literature on anthropogenic noise impacts on bats is focused on foraging and use of echolocation, I think the general  body of literature on noise impacts on wildlife in general is just a pertinent. My point being, regardless of whether the noise is impacting their echolocation abilities (or how much the species is using echolocation), extreme noise would still likely impact their behavior.

You have probably seen this, but here fairly recent literature review that might be helpful: 

[Francis and Barber 2013. A framework for understanding noise impacts on wildlife: an urgent conservation priority. Frontiers in Ecology and the Environment 11: 305–313.]

Your question requires a bit of clarification. When you talk about estimating SPL of bird calls, do you mean the received level (RL) of the calls at your microphone, or the source level (SL) as the bird produced the sound? SLs are typically given in dB SPL (or dB re 1 µPa) at a distance of 1 m from a source.

If you want SLs you first need to measure (or estimate) the RL and then apply an adjustment for the distance between the bird and your microphone, since RL falls off logarithmically with distance from the source.

In order to get actual RLs from a recording, you need to have calibrated (or more properly “characterized”) recording equipment so that you know the scaling relationship between the digital sample values in the audio recording and actual sound pressure at the microphone, measured in micropascals (µPa). There are two approaches commonly used to obtain this calibration info. The first “stepwise” calibration approach relies on knowing the way in which the signal is transformed at three stages: (1) the microphone, which converts pressure variations into voltage variations, (2) the preamplifier, which amplifies the weak electrical signal from the mic, and (3) the digitizer (analog-to-digital converter) which converts the continuous voltage waveform into a series of digital audio samples (typically 16- or 24-bit integers).

The second approach is to treat the entire recording system (mic + preamp + digitizer) as a “black box” and just determine what the scaling relationship is between the audio sample values and the sound pressure (in µPa) at the microphone. Usually this is done by making a recording of one or more calibration signals (usually pure tones) while also measuring the level at the microphone with a sound level meter.

These approaches are summarized in the attached Powerpoint slides.

Beginning in Raven Pro 1.5 (currently in a beta version) you can enter calibration information if you have it available to get true SPL values out from the measurements. Without that calibration info, dB measurements in Raven can be used for making relative comparisons between sounds in the same recording (or in different recordings if they were made with the same equipment and gain settings), but not to infer actual dB SPL.

One caveat to keep in mind is that if you change gain settings during a recording or between recordings, you need to know what the gain setting was at every time, and what this gain setting corresponds to in terms of dB of amplification.

Some of the occupancy models that account for both false positives and false negatives might be useful, e.g.

Accounting for false-positive acoustic detections of bats using occupancy models

By:Clement, MJ (Clement, Matthew J.)[ 1 ] ; Rodhouse, TJ (Rodhouse, Thomas J.)[ 2 ] ; Ormsbee, PC (Ormsbee, Patricia C.); Szewczak, JM (Szewczak, Joseph M.)[ 3 ] ; Nichols, JD (Nichols, James D.)[ 1 ]

JOURNAL OF APPLIED ECOLOGY

Volume: 51

Issue: 5

Pages: 1460-1467

DOI: 10.1111/1365-2664.12303

Published: OCT 2014

Hi Patricia,

I do not have yet papers to advise you, but I would be interested in following some of your work : I try to link bioacoustic patterns to the medium where they are emitted...

Sincerely,

Hervé glotin http://sabiod.org

I think that was caused by the evolution and the habitat selection.

I’ve never used the Olympus DM 650. I’m using Zoom H4n, to record birds (zebra finch and great tit). I think it’s a similar product. I’m pleased with this recorder. It’s easy to use, good quality and not too expensive.

But when I have to make passive recording in field condition I’m using SongMeter 2 from Wildlife Acoustic. It’s a weatherproof system which can be programed to record for example 2 hours after sunrise each morning. The battery life is really good (several weeks or mouth depending on the program).

Hope this can help.

This sounds like a long-range communication call. These are often used while moving fast in the  dense canopy, and when pair or flock members are separated by the distance of 10-15 m at least. 

Dear Tullio,

There is no magic bullet that will let you detect every type of signal.  For general data exploration, I'd suggest the following software packages:  xbat (Matlab program written by Harold Figueroa from Cornell - nicely done, no longer supported and documentation is supposed to be a bit on the slim side),. Cornell's Raven, Osprey (Matlab program written by Dave Mellinger at Oregon State that has some nice annotation features), or Triton (Matlab program written primarily by Sean Wiggins at Scripps Institution of Oceanography that is designed for very large datasets and has a nice compressed spectrogram function [some of the other packages have this as well]).  There are other packages out there as well.  For detecting calls, it really depends upon your soundscape and the calls you are interested in as to how easy that is to do.  The Teager energy detector was used by Kandia and Stylianou (2006) for detecting odontocete echolocation clicks and should work well for snapping shrimp; we developed it independently and while we never published as Kandia & Stylianou beat us to the punch our description can be seen in some our papers (look at Soldevilla et al. 2008 or Roch et al. 2011).  

I haven't done work on fish calls, but I would expect that for pulsed calls you would want to key in on some aspect of the frequency range and pulse rate. 

If you just want a general signal detector, you could look at the work of Erbe and King (2008), they designed a signal detector that is very easy to implement and can do a pretty nice job.

Best of luck in your research - Marie

Erbe, C., and King, A. R. (2008). "Automatic detection of marine mammals using information entropy," J. Acous. Soc. Am. 124(5), 2833-2840.

Kandia, V., and Stylianou, Y. (2006). "Detection of sperm whale clicks based on the Teager-Kaiser energy operator," Appl. Acous. 67(11-12), 1144-1163.

Roch, M. A., Klinck, H., Baumann-Pickering, S., Mellinger, D. K., Qui, S., Soldevilla, M. S., and Hildebrand, J. A. (2011). "Classification of echolocation clicks from odontocetes in the Southern California Bight," J. Acous. Soc. Am. 129(1), 467-475.

Soldevilla, M. S., Henderson, E. E., Campbell, G. S., Wiggins, S. M., Hildebrand, J. A., and Roch, M. A. (2008). "Classification of Risso's and Pacific white-sided dolphins using spectral properties of echolocation clicks," J. Acous. Soc. Am. 124(1), 609-624.

If you need sound level meter Class I as B&K but much cheaper, you can try Svantek products on page http://svantek.com/sound-meters-analysers.html. If you need usual FFT analysis SVAN 979 is your option, if you need higher frequncies 977 could be useful and if you need something small 971 is fantastic but with third-octave band analysis only (as I know). Transfer of data into PC is standard so you can use it for any usual data acquisition.

(I am not a reseller of any product :-))

So this way I can only wish You luck in Your further research.

       Best regards

        Chris

to evaluate a possible impact of noise on animals the A curve is misleading as it is based on the human hearing sensitivity curve. No weighting, or a weighting tailored on the sensitivity curve of the target species should be used instead. However, I strongly suggest to perform full range 1/3 octave analysis and possibly consider the percentile distribution of energy vs frequencies to have an idea of the possible impacts.

Not only "levels" should be considered; the spectral structure of the noise and its distribution in time (e.g. continuous, intermittent, random, ....) are important as well.

I agree with Charles Henry about anatomical validation. In our lab we have modified a GoPro action camera (which records at 240 frames per second and fantastic resolution) with macro lenses and have been getting incredible "high speed" video of echolocating bats for a fraction of typical high speed camera costs. Contact me directly if you'd like more information on this setup--all the components are available commercially from various companies.

Dear Mr. Schlindwein

1. It was about generation not propagation.

2. Why these subhertz mechanical waves would not propagate in gas or liquids? The initial amplitude is low enough not to generate nonlinear dissipation mechanisms and the speed of sound is the same in a medium, independent of wavelength. It is well known that elephants (and wales, eg: sperm wales) generate (using large resonating cavities) low freg. vibrations that succeed to propagate incredibly far.

Looking at the mechanical waves' propagation eq (source, Wikipedia):

Feynman[2] derives the wave equation that describes the behavior of sound in matter in one dimension (position x) as:

{ \partial^2 p \over \partial x ^2 } - {1 \over c^2} { \partial^2 p \over \partial t ^2 } = 0 ,

there is no apparent reason for these waves not to propagate in fluid media.

Hi Christina, I have been working with bottlenose dolphin burst pulsed sounds during the last years and I understand your need to find to define the differences between sounds. The existing literature is vague because burst pulses have been traditionally discussed in terms of their sonic properties and is difficult to measure when calls are classified solely by ear or acoustic features. Terms such as “screeches,” “gulps”, “brays”, "barks", “quacks”, “yelps”, and more… commonly used to describe and distinguish burst pulsed sounds can result in misleading conclusions, as they primarily describe the subjective impressions experienced by human listeners. Additionally, the way the vocalizations are analyzed, and the authors tendency to split or lump, also affects the interpretation of repertoire size. For example, a bray is formed by two different types of burst pulsed sounds however a bark is only one... In one of my studies I have classified the burst pulses based on the duration of the sound (in the oscillogram), in other studies I also measured the Interclick interval or interpulse interval manually (for this I have used a software called soundruler).

As a start, I suggest you to form classes of burst pulsed sounds on the basis of structural characteristics (for example I used the duration), this would lead to more meaningful comparisons between categories.

Good luck!

Are you also interested in the studies on the effects of noise on animals, especially frogs in China? May exchange our results with your study.

As far as bottlenose dolphins go:

The following paper may help answer your question.

it also appears that male bottlenose dolphins can experience age-related hearing loss.

Dr. Darlene Ketten (now working at the Centre for Marine Science and Technology, Curtin university of Technology, Perth Australia) has done some work in this area and might be able to help further if you contact her directly.

even though it sound's ridiculous these details are considered confidential. so, the only place you can find solution for this in EC office in belgium

First, let's get a question of terminology taken care of: to calibrate something, you need to do two separate processes. These are characterization and adjustment. Characterization is the act of finding what a device's response to a known stimulus or signal is. Adjustment is the act of changing the response of the device under test (or DUT). Calibration is the term that refers to the process of characterizing, adjusting, and re-characterizing a DUT so that its response is as close to perfect as you can make it. In your case, I don't imagine that you can change the response of the hydrophone or the recorder, so we'll just call what you want to do characterization, and use the data from the characterization to adjust your data via post-processing in software.

There are two ways to do this for your system: the first is to characterize the hydrophone and recorder separately, while the second is to characterize them together. If you do them separately, then you'll have data on both devices in case they're used with other devices in the future. If you do them together, then you'll have a faster method to get data on the setup you currently have. The choice is yours, and I'll describe the latter method first.

One aside first, though: the limit here is equipment. You'll need a calibrated hydrophone (hereinafter known as the standard hydrophone) that can give you a well-known sensitivity that is relatively flat across its response bandwidth, and you'll also need an underwater sound source, like an underwater speaker, or a speaker that's leaning up against a bucket or tank of water.

The first thing to do is set up your hydrophone (the 96-MIN) and recorder to make your recording. Set up the standard hydrophone so that its output is going into an oscilloscope. The hydrophones should be co-located at a distance of at least 10 cm from your sound source. Send a 1 kHz sine wave into the speaker at a variety of different levels (I usually do 1, 2, 5, 10, 20, 50, 100, 200, and 500 mVrms, and 1, 2, 5, and 10 Vrms). At each level, write down what amplitude the oscilloscope shows from the standard hydrophone. What you have now is a response table for the sound source, where the abscissa is in source voltage, and the ordinate is standard hydrophone voltage. The standard hydrophone will have a sensitivity in either V/µPa or dB re 1V/µPa. You use this number to convert first to µPa, and then to dB re 1 µPa, which will then give you the sound source SPL at each input voltage. When you're done with this process, repeat it for at least three other frequencies across the range of frequencies that you're interested in recording. Take the recording, and bring it into Raven. Once there, you'll only want to use the waveform view. Put your cursor over the portion of the waveform that's at the amplitude that you want, and then take down the minimum and maximum displayed at the bottom of the Raven window. These are the values from the sound file in Least Significant Bits (LSBs). The conversion factor that you need is LSB/µPa. To get this, graph a line with SPL as the abscissa and LSBs as the ordinate for each frequency. You can then use the line fit of your choice to generalize a dynamic response function for each frequency, and possibly one of the form H(A,f) for the entire response surface.

Similarly, if you want to characterize the response of the hydrophone and recorder separately, repeat the process as above, but use the oscilloscope for both the standard hydrophone and the DUT hydrophone, and then use the voltage source as above, but directly connected to the recorder's input. You'll get the same data out from the recorder, but instead of being in LSB/µPa, it will be in LSB/V, and the DUT hydrophone's output will give you V/µPa. Thus, if you multiply the two, you'll get the same LSB/µPa that you had above. These conversion units are all known as sensitivities, and can be used outside of the hydrophone/recorder system if you interface with anything else.

It is not clear whether you want a datalogger, which can be moored or just a recorder to use on a boat.

Dataloggers:

The Loggerhead is not ideal, at least in the standard version, exactly because it can only record very short (seconds or subseconds) at a time at sampling rates above 80 kHz. Furthermore, the standard hydrophone has a 40-50 kHz cut-off frequency. You should contact David Mann directly (on the Loggerhead web-page) and ask him for possibilities.

If you are only interested in knowing whether there are dolphin clicks or not and not really care about the signal characteristics, you might be able to use the Loggerhead as it is. Standard config is about 6000 US$

Wildlife Acoustics sell a high-frequency logger (SM2M-HF, I think), which can make full bandwidth recordings with a 300+ kHz sample rate. Price is around 10.000 US$ as I recall. That should do the job.

Recorders.

Simplest option is to use a laptop and a suitable A/D-converter board. Check the PamGuard software (freeware) and find a National Instruments A/D converter which can operate at least at 500 kHz per channel. If you only need one, you can get these fairly cheap. Look out for second hand versions to really save money.

If you are only interested in the click time information, the simplest and maybe also cheapest solution is to buy a click detector, such as the one sold by ETEC (etec.dk, ~900 Euro), which will trasform the ultrasonic clicks into the audible range and make it possible to record them on any standard audio-recorder, which you can buy for 100 $ or less.

Best regards

Jakob

I think the search terms you are looking for are "Granular Synthesis". You should find a lot of software to produce sounds for bioacoustics using this technique.