GB2535167A

GB2535167A - Audio signal processing apparatus, client device, system and method

Info

Publication number: GB2535167A
Application number: GB1502132.2A
Authority: GB
Inventors: Gosling Steve; Peckham Reuben; Coles Dave
Original assignee: 24 ACOUSTICS Ltd
Current assignee: 24 ACOUSTICS Ltd
Priority date: 2015-02-09
Filing date: 2015-02-09
Publication date: 2016-08-17
Anticipated expiration: 2035-02-09
Also published as: GB2535167B; GB201502132D0; WO2016128729A1

Abstract

An apparatus monitors sound signals and provides remote access to acoustic parameters generated from the monitored sound signals for graphical display. The apparatus comprises a receiver for receiving an audio signal from a microphone and an audio signal processor 16 to process the audio signal in accordance with a plurality of threads 20,22. Each thread comprises capturing a predetermined duration of the audio signal, generating a sampled digital version of the signal, and calling a next of the plurality of threads to capture a subsequent predetermined duration of the audio signal. Each of the threads are called in turn, one after the other in a cycle from a first to a last, which then repeats with the effect that the audio signal is continuously processed. The processor analyses each of the sampled digital signals to generate predetermined acoustic parameters for each of the predetermined durations, and to store the acoustic parameters with respect to an indication of a temporal period of the audio signal to which the acoustic parameters were generated. A client device may access the acoustic parameters and encoded compressed audio files via a network 6. The apparatus may monitor noise from a wind farm.

Description

AUDIO SIGNAL PROCESSING APPARATUS, CLIENT DEVICE, SYSTEM AND METHOD BACKGROUND

Field of Disclosure

The present technique relates to audio signal processing apparatus and systems for acoustic monitoring and methods of processing audio signals. Embodiments of the present technique can provide an arrangement in which acoustic parameters can be accessed remotely and can be used to generate a graphical representation of monitored audio signals for remote access.

Description of Related Art

The "background" description provided herein is for the purpose of generally presenting the context of the disclosure. Work of the presently named inventors, to the extent it is described in this background section, as well as aspects of the description which may not otherwise qualify as prior art at the time of filing, are neither expressly or impliedly admitted as prior art against the present disclosure.

There are various scenarios and applications in which it is necessary to provide some monitoring of noise levels produced for various sources. For example, building work and land based oil production facilities may be located within or near residential areas and part of a condition for these activities may be that noise emissions are reduced or restricted to a limit defined by local authorities such as for example a mean average or a maximum level. Another example is the deployment of wind farms comprising either a single or a plurality of wind turbines. A phenomenon known as "amplitude modulation" which results from an action of the rotor blades of the wind turbine interacting with the wind itself can produce a disturbing noise.

It is known to provide in situ noise monitoring of noise sources in order to determine that the noise levels are complying with a predetermined requirements. However known techniques require the disposal of a noise monitoring apparatus at a site and typically the noise monitoring apparatus is manually operated or automated in a basic manner in order to confirm that the noise levels comply with predetermined requirements. As will be appreciated there is a desire to provide improvements for such arrangements.

SUMMARY OF THE DISCLOSURE

According to the present technique there is provided an apparatus for processing audio signals to monitor sound signals and to provide remote access to acoustic parameters generated from the monitored sound signals. The apparatus for processing the audio signals comprises a receiver for receiving an audio signal from a microphone representing the sound signals received by the microphone and an audio signal processor configured to process the audio signal in accordance with a plurality of threads. Each thread comprises capturing a predetermined duration of the audio signal, generating a sampled digital version of the captured predetermined duration of the audio signal, and calling a next of the plurality of threads to capture a subsequent predetermined duration of the audio signal. Each of the threads are called in turn, one after the other in a cycle from a first to a last, which then repeats with the effect that the audio signal is continuously processed by the audio signal processor to generate, for each of the predetermined durations, which are contiguous in a time, the sampled digital versions of the audio signal. The audio signal processor is configured to analyse each of the sampled digital versions of the audio signal to generate predetermined acoustic parameters for each of the predetermined durations, and to store the acoustic parameters with respect to an indication of a temporal period of the audio signal to which the acoustic parameters were generated.

According to embodiments of the present technique the audio signal processing apparatus is configured to monitor continuously a noise source and generate acoustic parameters corresponding to each of a plurality of temporal periods into which the continuous sound signal is divided which can be monitored with respect to the acoustic parameters to identify whether the sound source exceeds or meets the predetermined parameters. This arrangement is achieved by providing a plurality of threads for example two threads, in which one calls the other each thread being arranged to generate an uncompressed digital sample version of the audio signal which can then be processed in order to generate the acoustic parameters with respect to which the sound source is being monitored.

According to one example embodiment the audio signalling processing apparatus includes a network interface. The network interface provides an arrangement in which a remote client terminal or computer can access the acoustic parameters via an internet connection for display on the computer so that the audio signal source can be monitored remotely.

According to another example, audio files corresponding to each of the predetermined durations of the audio signal are converted into compressed files. The network interface is configured to provide the compressed audio files to the computer via the internet connection so that if a sample corresponding to the predetermined duration of the acoustic signal exceeds predetermined parameters, the sound corresponding to that audio file can be accessed via the internet connection by the computer thereby determining whether the sound which caused the acoustic parameters to be exceeded was emitted by a particular sound source or by a spurious sound source. In one example the compressed audio files are generated from the audio files of the captured predetermined duration of the audio signal and provided via the network interface to the computer, and subject to predetermined conditions being satisfied the audio file of the captured predetermined duration of the audio signal is also stored for access via the network interface to the computer.

Further aspects and features of the present technique will be defined in the appended claims.

The foregoing paragraphs have been provided by way of general introduction, and are not intended to limit the scope of the following claims. The described embodiments, together with further advantages, will be best understood by reference to the following detailed description taken in conjunction with the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

A more complete appreciation of the disclosure and many of the attendant advantages thereof will be readily obtained as the same becomes better understood by reference to the following detailed description when considered in connection with the accompanying drawings wherein like reference numerals designate identical or corresponding parts throughout the several views, and wherein: Figure 1 is a schematic representation of a deployment of a sound monitoring apparatus on a wind farm according to the present technique; Figure 2 is a schematic block diagram of audio monitoring system including an audio monitoring apparatus according to the present technique; Figure 3 is a part graphical, part process representation of a process performed by an audio signal processing apparatus showing in Figure 2 to form a plurality of threads for processing the audio signal; Figure 4 is a schematic block diagram of processes performed by the audio signal processor according to the present technique; Figure 5 is a schematic block diagram of an audio processing device shown in Figure 4; Figure 6 is a schematic block diagram of an audio statistics calculator shown in Figure 4; 30 Figure 7 is an example flow diagram illustrating an example embodiment of the present technique; and Figures 8 is a graphical plot of an expanded 10 minute sample with each point representing a ten second average with modulation depth on a primary y-axis and modulation depth as measured in accordance with the RUK method (on band-filtered data) is shown against the secondary y-axis; Figure 9 is a graphical plot of ten minute average sample values of pressure levels on the primary y-axis and modulation frequency on a secondary y-axis; Figure 10 is a is a graphical plot of showing the audio samples presented in Figure 9, with a specific point in time selected and the corresponding audio file sewed to the client; Figure 11 is a is a graphical plot showing another example corresponding to the graphical plots of Figures 8, 9 and 10 in which demolition noise is measured, with audio samples provided by as one minute average values for each sampling point on one plot and one hour average values shown as bars on top of theonel minute average values.

Figure 12 is a graphical plot showing the results presented in Figure 11, with a specific one minute average value selected and the audio sample playing back to the client through via internet.

DETAILED DESCRIPTION OF THE EMBODIMENTS

An example embodiment of the present technique is illustrated by a diagram shown in Figure 1. Figure 1 provides an example in which an audio signal processing apparatus according to the present technique 1 is deployed to monitor sound emissions from a wind farm comprising a plurality of wind turbines 2, which generate electricity from wind flowing past turbine blades 3.

A phenomenon known to those skilled in the art as "amplitude modulation", as explained above, can be produced by the turbine blades 3 in accordance with particular conditions as a result of the action of the wind on the turbine blades 3. This amplitude modulation can be a nuisance to residents living near the wind farm. Accordingly it may be desirable to monitor the sound produced by the wind farm in order to determine whether or not any nuisance noise has been produced by the wind farm as a result for example of amplitude modulation or whether this is in fact as a result of some other spurious noise source such as an aeroplane passing overhead.

As shown in Figure 1, the audio signal processing apparatus includes one or more microphones 4 which detect sound generated by the wind farm and converts the detected sound into audio signals which are received by the audio signal processing apparatus 1. As will be explained in the following paragraphs, the audio signals are monitored by the audio signalling processing apparatus and from which data representing predetermined acoustic parameters corresponding to each of a plurality of predetermined durations or sections of time of the audio signal are produced which can be accessed via an internet connection, such as the world wide web, 6 by a computer 8. The computer can retrieve the audio files generated by the audio signal processing apparatus and can also retrieve monitored acoustic parameters, which can be displayed in graphical form by a client terminal as for example shown in Figures 8 to 12. Furthermore, according to the present technique, the audio signal can be converted into a compressed file and accessed via the internet 6 by the computer 8 so that this sound can be reproduced for a user of the computer 8 remotely. In one example, the compressed file can be generated for each of the predetermined durations or sections of time of the audio signal, or the compressed file can be generated under the condition that a sample of the section of time of the audio signal causes the predetermined acoustic parameters to exceed predetermined conditions. Then, to confirm whether or not sound which caused the acoustic parameters to be exceeded was produced by the wind farm or was in fact produced by spurious noise source, a client application program running on the computer 8 can access the compressed audio file corresponding to the section of time of the audio signal. In accordance with a conventional arrangement, this can be achieved, by downloading or triggering a file transfer of a copy of the audio file from the audio signal processing apparatus to the computer under the control of the client application program. Furthermore the captured predetermined duration of the audio signal for which the monitored acoustic parameters exceed predetermined trigger conditions can also be stored for access via the internet from a remote computer so that a highest fidelity version of the audio signal can be played back and reviewed to determine a reason for the monitored acoustic parameters to be exceeded or allow calculation of additional acoustic parameters.

According to the present technique the audio processing apparatus functions by continuously recording audio signals in predetermined durations or short segments (for example 10 seconds long) and extracting acoustic parameters by processing each audio file after capture. The capture of gapless audio files, each 10 seconds in duration, is achieved in the following manner: * A single, continuous audio stream from the audio interface is opened.

* An audio capture "thread" (referred to here as Thread 1) calls another audio capture thread (here referred to as Thread 2) with a delay of precisely 10 seconds.

* Thread 1 then retrieves precisely 10 seconds of audio.

* After capturing 10 seconds of audio, Thread 1 saves the audio in an uncompressed format (currently 24-bit WAV) and passes the filename of the saved audio to a signal processing algorithm.

* While Thread 1 is saving the captured audio to file and passing it to the processing algorithm, Thread 2 is started (having been called with a 10 second delay).

* The process is then repeated from Thread 2 to Thread 1, i.e. Thread 2 calls Thread 1 with a delay of precisely 10 seconds, Thread 2 captures 10 seconds of audio, Thread 2 saves the audio and sends it to the processing algorithm while Thread 1 starts again (after the 10 second delay).

Figure 2 discloses a more detailed representation of the audio signalling processing apparatus 1 shown in Figure 1. In Figure 2 the one or more microphones 4 feed audio signals to a pre-amplifier 10, which are then fed to an audio interface 12 under the control of a controller 14.

The audio interface 12 and preamplifier 10 receive and amplify respectively the audio signal which are received from the one or more microphones 4, which is then fed to a signal processor 16. Also forming part of the audio signal processing apparatus shown in Figure 1 is a network interface 18 which can be accessed via an internet 26 so that a computer 8 can access information generated by the signal processor from the audio signal under the control of the controller which can then be accessed via a communications network such as the internet 6 at a remote terminal such as the computer 8, as explained above.

According to the present technique the signal processor 16 in the audio signal processing apparatus 1 is configured to monitor continuously the sound signal detected by the microphone 4 so that the audio signal is continuously monitored. To this end, the signal processor 16 employs a series of "threads" which each perform a sequence of predetermined actions on the audio signal to generate uncompressed digital sample audio files. Each thread is arranged to call another thread and each of the threads therefore is arranged in sequence which is cycled through in order to generate continuously uncompressed digital audio files. In one example, the digital audio files representing the captured digital version of the predetermined duration of the audio signal could be for example WAV files of the sound source which can then be processed by the signal processor 16.

In one example the next thread in the sequence of threads is called at the end of the thread.

However advantageously, the next thread is called as a first process step of the current thread with a delay equal to exactly the predetermined recording period of the thread. This is to ensure that a next thread is called by the processor exactly when the thread is needed to provide a continuous capturing of the audio signal. Since the processor must execute other tasks during the thread there is otherwise a possibility that a variation in the processing may cause a length of time for performing the thread to vary and therefore potentially over-run a time when the next thread needs to be called in order to ensure the continuous capturing of the audio signal. Accordingly calling the next thread with a delay equal to a fixed and predetermined recording length of the thread ensures that the audio signal is continuously captured with no time drift.

A schematic representation of the processing of the sound signal to produce the audio files according to two threads is represented in Figures 3a, 3b and 3c. In Figure 3a a graphical plot is shown representing amplitude with respect to time of an audio signal. As shown in Figure 3b a first thread, THREAD 1, is shown, which captures a first predetermined period P of sampled digital audio 300. The first task performed in the first thread is to call a next thread in the sequence, THREAD 2, with a predetermined delay period D, which is equal to a total allocated for recording within a thread. As mentioned above, this is so that any processes performed by the processor during the thread do not prevent it from calling the next thread on time so that a continuous process of capturing the audio signal is not interrupted. As shown in Figure 3c, after the delay period D = ti, the next thread is called, THREAD 2, a first process step of which is to call the next thread in the sequence, THREAD 1. After capturing the next predetermined duration of the audio signal 302, and following a delay D, the next thread, THREAD t, is performed to capture the next predetermined duration of the audio 304. In this way a plurality of threads can be arranged to process continuously the audio signal generated for each of the predetermined durations P, which correspond to the predetermined durations of the captured digital audio, which has a sampling rate at this point of, for example, 48kHz. In one example the predetermined durations of the audio signal are formed into audio files, which provided a highest fidelity of the audio signal captured by the audio processing apparatus. Further detail of the operation of the signal processor 16 shown in Figure 2 if presented in Figure 4.

As shown in Figure 4, the audio signal is processed by the signal processor 16 by calling a plurality of threads 22 in sequence thereby cycling through each of the plurality of threads. As explained above with reference to Figure 3, each of the threads generates an audio file 24. As shown in Figure 4, five audio files are produced but this is just an illustration of audio files which would be continuously produced. In one example, the audio files 24 are WAV files generated by performing analogue to digital conversion at a sample rate of 48 kHz. The audio files in one example are WAV files, which are continuously generated. However the audio files are representative of digital sampled files, which are continuously processed by the audio signal processor 16.

In one example the captured predetermined durations of the audio signal, which are formed into audio files are uncompressed or loss-less compressed to preserve the fidelity of the captured audio signals. As will be appreciated however some compression may be applied to the audio files, although as will be explained in the following paragraphs, the audio files 24, should as far as possible provide a fidelity which can ensure that the captured audio signal can be verified in terms of its acoustic content. For example the audio files 24 may be substantially uncompressed, so that the acoustic information is preserved to the extent that acoustic parameters can be determined correctly. Therefore the term uncompressed or substantially compressed may be interpreted to the extent that the acoustic fidelity is retained to the extent that Class 1 and Class 2 parameters can be calculated according to the annexed defined British Standard BS EN 61672-1, part 5.2.2.

As shown in Figure 4, the audio signal processor 16 comprises an audio processor unit 30 which is configured to receive the audio files 24 at an input 33. The audio signal processor unit 30 feeds the audio files 24 to an audio pre-processor unit 31, which pre-processes the audio file which is then fed to a re-sample audio unit 32, which processes and re-samples the captured digital audio samples of the audio file. In one example this is achieved by down sampling each file to generate a form, which processes the audio file in for example 100 millisecond RNIS samples of the audio signal. The audio file is then fed to an acoustic parameter calculator 34 which, in one example, determines acoustic parameters in accordance with a British standard BS EN 61672-1. According to this example, as will be explained in more detail below, the sampled version of the audio files 24 is used to generate acoustic statistics calculating an LAeq and Lmax values or other predetermined measurement parameters for the audio signal in each of the predetermined durations. The acoustic parameters are then passed to a data store 40 under the control of the controller 36 and stored with respect to the predetermined duration of the audio file for which the acoustic parameters were generated. The acoustic parameters may be generated at any predetermined sampling rate within the predetermined generation of the audio file.

In accordance with one example embodiment of the present technique, each of the received audio files which are substantially uncompressed digital samples for example WAV files are received by a compressed audio file generator 38, which generates a compressed version of the WAV audio file which is also stored in the data store 40. In one example, the compressed audio file is always generated whereas in other examples the compressed audio file is only generated if the acoustic parameters satisfy predetermined conditions for example exceeding an average amplitude for a predetermined period. Generally, the original substantially uncompressed audio files 24 are not stored in the data store 40 so that once the uncompressed audio signal files have been processed by the acoustic parameter calculator 34 the audio files are discarded. However, in one example if the acoustic parameters satisfy predetermined conditions then the uncompressed audio files 24 are also stored in the data store 40.

In accordance with the present technique and as explained above, the controller 36 shown in Figure 4 is arranged to provide access via the network interface 18 to the data store 40.

According to the present technique, a client operating a computer 8 may access the internet 6 via the network interface 18 and receive the acoustic parameter stored in the data store 40 and/or the compressed audio file stored in the data store 40 corresponding to the acoustic parameters. Accordingly, the acoustic parameters may be monitored remotely and furthermore the sound corresponding to the acoustic parameters may be accessed by retrieving the compressed audio file for example an open source format such as.00G.

As shown in Figure 4 an audio processor unit 30 processes the received WAV file in accordance with a controlled arrangement which may be user selectable. The audio processor unit 30 may include a series of standard filters to calculate noise parameters. The audio processor 30 may include (if specified by the user): * Single octave band spectrum is calculated * Third octave band spectrum is calculated * An A-weighting filter or frequency weighting filter, is applied to the audio signal * A time-weighting filter is applied to the audio signal The re-sampling unit 32, may be used to form the discrete samples of the audio data into 100ms samples. The acoustic parameter calculator 34 may calculate according to the above mentioned British standard BS EN 61672-1 Leq, Lmax, and other statistics such as peak sample value, mean duration above a predetermined threshold.

Once relevant parameters have been calculated, the audio file can be discarded or stored on the data store in either a compressed or uncompressed format. It is possible for the choice of keeping/discarding audio to be made by whether predetermined criteria have been met.

If compression is used, a significant duration (e.g. several months) of audio can be stored in the data store. These audio clips can be remotely requested through the web server which allows a user to view noise levels on a graph and listen to audio by selecting a data sample of interest as will be explained shortly with reference to Figures 8 to 12.

One example of the audio pre-processor unit 31 shown in Figure 4 is shown in more detail in Figure 5. As shown in Figure 5 the audio pre-processor unit 31 comprises a single octave filter 50, a third octave filter 52, and an A weighting or frequency weighting filter 54. The octave filter 50, the third octave filter 52, and the A weighting filter 54 serve to filter and pre-process the audio signal. In accordance with the present technique, the parameters for each of these filters 50, 52, 54 may be selected by the controller 16 so that the audio signal may be filtered to identify a particular characteristic of noise which the audio processing apparatus is monitoring.

Figure 6 provides a more detailed example of the re-sampling audio unit 32 shown in Figure 4.

As shown in Figure 6 the re-sampling audio unit 32 comprises a first processing unit 58 which forms 100 millisecond samples from the uncompressed digital data. The 100 millisecond samples are then fed to a time weighting filter 60, which performs time domain weighting of the samples.

As explained above, the acoustic parameter calculator 34 then calculates predetermined acoustic parameters.

The table set out below provides a summary of the audio processing which is performed by the audio signal processing apparatus. For each stage, a description is given along with the format of the audio data files for each stage and the format of the files outputs from that stage.

Description of Process Audio Data format Outputs Processed by Example Element Octave and third octave band levels are calculated from the raw audio waveform using an FFT based calculation (akin to that utilised by real-time signal analysers) Uncompressed WAV file (typically 24 bit, 48 kHz) Octave and third octave band levels Audio pre-processor unit 31 An A-weighting filter Uncompressed WAV file (typically 24 bit, A-weighted audio file Audio pre-processor unit 31 is applied to the audio file in the time-domain 48 kHz), A-weighted The audio file is sampled into 100 ms chunks by taking the root-mean-square (see Section 3.9 of BS EN 61672-1) A-weighted 100 ms samples, with amplitude in calibrated dB 100 ms LAeq data points Audio processor 30 and re-sampling audio unit 58 The 100 ms data are time-weighted using the methodology detailed in NORDTEST Method NT,', C01.: 112 A-weighted and time-weighted 100 ms samples, with amplitude in calibrated dB 100 ms frequency and time-weighted data points Audio processor 30 and time weighting filter 60 The average LAeq over the sampling period (for example 1 minute) is calculated by taking the logarithmic average of all 100 ms LAeq values A-weighted 100 ms samples LAeq,T (where Tis the time-period) Acoustic parameter calculator 34 The LAmax,f is calculated by taking the maximum of the time-weighted 100 ms within the period I' A-weighted, time-weighted 100 ms samples LAmax,fT (where Tis the time-period) Acoustic parameter calculator 34 Statistical values (such as L10, L50 and L90) are calculated from the time-weighted 100 ms data. These are calculated by taking the relevant percentile from all the time-weighted 100 ms samples within t period T A-weighted, time-weighted 100 ms samples Statistical parameters, e.g. LA90,10min (where Tis the time-period) Acoustic parameter calculator 34 Further parameters as specified by user Acoustic parameter calculator 34 Figure 7 presents a flow diagram summarising the operation of the audio signal processing apparatus 1 shown in Figures 1 to 6 in accordance with the present technique. Figure 7 is summarised as follows: S2: Receive an audio signal from a microphone representing sound signals received by the microphone.

S4: Process the audio signal in accordance with a plurality of threads, each thread comprising * calling a next of the plurality of threads to capture a subsequent predetermined duration of the audio signal to start with a predetermined delay, * capturing a predetermined duration of the audio signal, and * generating a sampled digital version of the captured predetermined duration of the audio signal. *

Each of the threads are called in turn, one after the other in a cycle from a first to a last, which then repeats with the effect that the audio signal is continuously processed by the audio signal processor to generate, for each of the predetermined durations which are contiguous in a time.

56: Each audio file is processed by an audio signal processor to filter and to shape the received audio signal file.

S8: The uncompressed digital audio signal is then processed and formed into 100ms audio samples which is a lossless process and is therefore uncompressed.

S 10: The uncompressed 100ms samples are then used to generate one or more predetermined acoustic parameters according to defined statistics.

S12: A compressed version of each of the uncompressed audio files is then generated.

SI4: At decision point SI4 it is determined whether the predetermined acoustic parameters calculated in step S10 exceed one or more predetermined thresholds depending on the number of acoustic parameters being monitored. If one or more of the parameters do exceed the predetermined threshold then processing proceeds to step S16. Otherwise in step S18, the acoustic parameters are stored in a data store, with the compressed audio file to make the compressed audio file and the acoustic parameters available to be published to one or more client terminals via a network interface.

518: If the monitored acoustic parameters exceed a predetermined threshold then processing proceeds to step S 16 and the audio processing apparatus is arranged to store the uncompressed audio file along with the acoustic parameters for the predetermined period of the captured audio. The audio file is therefore stored with the acoustic parameters in a data store identifying the duration or period with respect to which the audio file was generated.

S20: The content of the data store comprising the determined acoustic parameters and audio files for each of the predetermined durations arc then made available via a network interface to a client terminal.

In one embodiment the audio signal processing apparatus may be operatively coupled via the network interface to a sewer, which may provide a cloud storage facility. The acoustic parameters, the compressed audio file and in accordance with a trigger event the audio files representing the predetermined duration of the captured audio signal in substantially uncompressed form may be stored on the server in accordance with a cloud storage arrangement. According to one example monitored acoustic parameters may cause a trigger event to occur when the parameters exceed one or more predetermined thresholds. This trigger event may be detected by the server as part of the cloud processing and an alarm signal sent to one or more client devices.

Figures 8 to 12 provides a graphical representation of an example of a graphical output presented on a computer 8 by a client accessing the data from the audio signal processing apparatus 1 via the internet. Figures 8 to 12 provide a graphical representation of an example of a graphical output presented on a computer 8 by a client accessing the data from the audio signal processing apparatus 1 via the internet.

Figure 8 shows one example where noise from a wind turbine site is measured. Each sample represents a 10 minute average in time. Overall sound pressure level is represented by the LA90 acoustic parameter and is shown on the primary y-axis. Modulation depth as measured in accordance with the RUK method (on band-filtered data) is shown on the secondary y-axis. Each 10 minute sample in Figure 8 can be selected individually and expanded to show sixty 10-second averages within the 10 minute period.

Figure 9 shows an expanded 10 minute sample with each point representing a 10 second average. Modulation depth is shown on the primary y-axis and the modulation frequency is shown against the secondary y-axis. Should the user wish to listen to the audio corresponding to a specific time, the point on the chart can be selected and audio requested using the "Download Selected Audio" button.

Figure 10 is a graphical plot of showing the audio samples presented in Figure 9, with a specific point in time selected and the corresponding audio file served to the client.

Figure 11 shows another example, where demolition noise is measured. 1 minute average values are shown as individual points on a line and 1 hour average values are shown as coloured bars on top of the 1 minute average values. Each point on the chart corresponding to a 1 minute average can be selected in order to request the corresponding audio recording.

Figure 12 shows the chart as described in Figure I I, with a specific 1 minute average value selected and the audio sample playing back to the client through the internet.

Annex 1: BS EN 61672-1 2003 has a definition for LAN: It also has the following definition for time-weighted level: 349 time-aver * equivalent canti twenty times the doriAg 4 -stated he's with a etan,ileld fro(' d ous Sound 1z vel e e o \ e' 1.V. iN* etri:n-ja wan}Mei- S Ps..{4}1:', this A-Vt*.1 *t ctE sound pre, taerig abtane rang. $:

time-weighted sound tevel * Ickenty hmed the ingerithrn to the base ten of the ratiO of a given ri;)c1A-meart-square sound pressure to the reference sound pressure, tootsteesti-sttuare sound pressure:bete9 obtained with a standard frequensy weighting and t& dad trine weighting.

not's, 1 Tend-wed)Med.&))))na ran& iledas,aa (4A NOTE Pot arno-nansanten teen)) n>dande latter aymnees;am alku, Lcs: ,t3 for eanneney weidahnt.)s A esti C and 1,(nei* weaghtingn h and S NOTE= narrndoia, A-actiehtga and talan-neadr)te; Sneed idee$, atarry.radheat;;)-{)rahe t reptCgi4nt.d,

-

) if p vi /A; iit Wht:rKf 6,<palet;,Ita4 t"^8'keee f(n hies 41/4414-14h4ing $: -zt*ery 'Nh ''f k, ,0**, Vs''* * ** "a.t t, * k* ',5.^ -.Or a.ks, (4, inteorai is..., the:isle et observe:ion); )he A-weiehund)eatenr:3-113ik:i.sound * Int) 5aVVIC.: primstatl:, In end:Mann rne nineenetot dt tea err()aarne:-a dg theIndere:am) tna) r sdned »48staeahortflt.f.

NOTh) 4 The earelne ha ranee I illunznah-sa:Pe osseesS 'radii-n(nee ennadah (1)

Claims

CLAIMSWhat is claimed is: 1. An apparatus for capturing and processing audio signals, the apparatus comprising a receiver for receiving an audio signal from a microphone representing sound signals received 5 by the microphone, an audio signal processor configured to process the audio signal in accordance with a plurality of threads, each thread comprising capturing a predetermined duration of the audio signal, generating a sampled digital version of the captured predetermined duration of the audio signal, and calling a next of the plurality of threads to capture a subsequent predetermined duration of the audio signal, wherein each of the threads are called in turn, one after the other in a cycle from a first to a last, which then repeats with the effect that the audio signal is continuously processed by the audio signal processor to generate, for each of the predetermined durations, which arc contiguous in a time, the sampled digital version of the audio signal, and the audio signal processor is configured to analyse each of the sampled digital versions of the audio signal to generate predetermined acoustic parameters for each of the predetermined durations, and to store the acoustic parameters with respect to an indication of a temporal period of the audio signal to which the acoustic parameters were generated.
2. An apparatus as claimed in Claim 1, wherein each thread comprises first calling the next of the plurality of threads with a start delay for the thread which is equal to the predetermined thread recording time, and then capturing the predetermined duration of the audio signal, and generating the sampled digital version of the captured predetermined duration of the audio signal, so that the audio signal can be continuously captured by the plurality of the threads called in the sequence.
3. An apparatus as claimed in Claim 1 or 2, comprising a network interface for providing access to the acoustic parameters and audio and the indication of the predetermined duration to which the acoustic parameters relate to one or more client terminals via a communications network.
4. An apparatus as claimed in Claim 1, 2 or 3, wherein the audio signal processor is configured to generate the sampled digital version of the captured predetermined duration of the audio signal as a substantially uncompressed sampled digital version of the captured predetermined duration of the audio signal.
5. An apparatus as claimed in any of Claims 1 to 4, wherein the audio signal processor is configured to generate a compression encoded data file representing the captured predetermined duration of the audio signal, to store the compression encoded data file in the data store, and to discard the sampled digital version of the captured predetermined.
6. An apparatus as claimed in Claim 5, wherein the audio signal processor is configured to compare the predetermined acoustic parameters with one or more thresholds, and if the predetermined acoustic parameters exceed one or more of the thresholds, to determine that a trigger event has occurred, and if the trigger event is determined to have occurred, then storing an audio file representing the predetermined duration of the audio signal for which the predetermined acoustic parameters exceeded the one or more thresholds.
7. An apparatus as claimed in any of Claims I to 6, wherein the network interface is configured to provide access to the compression encoded data file for each of the predetermined durations of the audio signal and in accordance with the trigger event the audio files representing the predetermined duration of the audio signal, stored in the data store, to a network for access by one or more client devices.
8. An apparatus as claimed in Claim 7, wherein the audio signal processor is configured, if the trigger event is detected to transmit an alarm signal to one or more client devices.
9. A method of processing audio signals, the method comprising receiving an audio signal from a microphone representing sound signals received by the microphone, processing the audio signal in accordance with a plurality of threads, each thread comprising capturing a predetermined duration of the audio signal, generating a sampled digital version of the captured predetermined duration of the audio signal, and calling a next of the plurality of threads to capture a subsequent predetermined duration of the audio signal, wherein each of the threads are called in turn, one after the other in a cycle from a first to a last, which then repeats with the effect that the audio signal is continuously processed to generate, for each of the predetermined durations which are contiguous in a time, the sampled digital version of the audio signal, analysing each of the predetermined durations of the audio signal to generate predetermined acoustic parameters for each of the predetermined durations, and storing the acoustic parameters with respect to an indication of a temporal period of the audio signal to which the acoustic parameters were generated.
10. A method as claimed in Claim 9, wherein each thread comprises first calling the next of the plurality of threads with a start delay for the thread which is equal to the predetermined thread recording time, and then capturing the predetermined duration of the audio signal, and generating the sampled digital version of the captured predetermined duration of the audio signal, so that the audio signal can be continuously captured by the plurality of the threads called in the sequence.
11. A method as claimed in Claim 9 or 10, comprising providing access to the acoustic parameters and audio and the indication of the temporal period to which the audio and acoustic parameters relate to one or more client terminals via a communications network.
12. A method as claimed in Claim 9, 10 or 11, comprising generating the sampled digital version of the captured predetermined duration of the audio signal as a substantially uncompressed sampled digital version of the captured predetermined duration of the audio signal.
13. A method as claimed in any of Claims 10, 11 or 12, the method comprising generating a compression encoded data file representing the sampled digital version of the captured predetermined duration of the audio signal, storing the compression encoded data file in the data store, and discarding the sampled digital version of the captured predetermined.
14. A method as claimed in Claim 13, comprising comparing the predetermined acoustic parameters with one or more thresholds, and if the predetermined acoustic parameters exceed the one or more thresholds, determining that a. trigger event has occurred, and if the trigger event is determined to have occurred, then storing an audio file representing the predetermined duration of the audio signal for which the predetermined acoustic parameters exceeded the one or more thresholds.
15. A method as claimed in any of Claims 10 to 14, comprising providing access to the compression encoded data file for each of the predetermined durations of the audio signal mid in accordance with the trigger event the audio files representing the predetermined duration of the audio signal, stored in the data store, to a network for access by on or more client terminals.
16. A method as claimed in Claim 15, comprising, if the trigger event is detected, transmitting an alarm signal to one or more client terminals.
17. A client device for providing access to acoustic parameters provided by an audio signal processor via a network, the client device being configured to access audio and acoustic parameters generated by an audio signal processing apparatus via a communications network, the audio and acoustic parameters having been generated by receiving an audio signal from a microphone representing sound signals received by the microphone, processing the audio signal in accordance with a plurality of threads, each thread comprising capturing a predetermined duration of the audio signal, generating a sampled digital version of the captured predetermined duration of the audio signal, and calling a next of the plurality of threads to capture a subsequent predetermined duration of the audio signal, wherein each of the threads are called in turn, one after the other in a cycle from a first to a last, which then repeats with the effect that the audio signal is continuously processed by the audio signal processor to generate, for each of the predetermined durations which are contiguous in a time, the sampled digital version of the audio signal, analysing each of the sampled digital versions to generate predetermined acoustic parameters for each of the predetermined durations, and storing the acoustic parameters with respect to an indication of a temporal period of the audio signal to which the acoustic parameters were generated.
18. A system for monitoring a. sound signal remotely, the system comprising an apparatus for processing audio signals, and a client terminal configured to receiver data from the apparatus for processing audio signals via a communications network, the data representing acoustic parameters detected from an audio signal representing the sound signal, the apparatus for processing audio signals comprising a receiver for receiving the audio signal from a microphone representing the sound signal received by the microphone, an audio signal processor configured to process the audio signal in accordance with a plurality of threads, each thread comprising capturing a predetermined duration of the audio signal, generating a sampled digital version of the captured predetermined duration of the audio signal, and calling a next of the plurality of threads to capture a subsequent predetermined duration of the audio signal, wherein each of the threads are called in turn, one after the other in a cycle from a first to a last, which then repeats with the effect that the audio signal is continuously processed by the audio signal processor to generate, for each of the predetermined durations, which are contiguous in a time, the sampled digital version of the audio signal, and the audio signal processor is configured to analyse each of the sampled digital versions of the audio signal to generate the data representing the predetermined acoustic parameters for each of the predetermined durations, and to store the acoustic parameters with respect to an indication of a temporal period of the audio signal to which the acoustic parameters were generated, and a network interface for providing access to the data representing the acoustic parameters and the indication of the predetermined during to which the acoustic parameters relate to the client terminal via 5 the communications network.
19. A system as claimed in Claim 18, wherein each thread comprises first calling the next of the plurality of threads with a start delay for he thread which is equal to the predetermined thread recording time and then capturing the predetermined duration of the audio signal, and generating the sampled digital version of the captured predetermined duration of the audio signal, so that the audio signal can be continuously captured by the plurality of the threads called in the sequence.
20. A system as claimed in Claim 18 or 19, wherein the audio signal processor is configured to generate the sampled digital version of the captured predetermined duration of the audio signal as an uncompressed sampled digital version of the captured predetermined duration of the audio signal.
21. A system as claimed in Claim 18, 19 or 20, comprising a server operatively coupled to the network interface, wherein the data representing the acoustic parameters are stored in the server for remote access by the one or more client devices.
22. A system as claimed in any of Claims 18 to 21, wherein the audio signal processor is configured to compare the predetermined acoustic parameters with one or more thresholds, and if the predetermined acoustic parameters exceed one or more of the thresholds, to determine that a trigger event has occurred, and if the trigger event is determined to have occurred, then storing an audio file representing the predetermined duration of the audio signal for which the predetermined acoustic parameters exceeded the one or more thresholds in the server.
23. A system as claimed in Claim 22, wherein the server is configured to provide access to the compression encoded data file for each of the predetermined durations of the audio signal and in accordance with the trigger event the audio files representing the predetermined duration of the audio signal, stored in the server, to one or more client devices.
24. A system as claimed in Claim 21, 22 or 23, wherein the server is configured, if the trigger event is detected to transmit an alarm signal to one or more client devices.