GB2429346A

GB2429346A - User-selectable limits in audio level control

Info

Publication number: GB2429346A
Application number: GB0605202A
Authority: GB
Inventors: Nicola Leotta
Original assignee: NEC Technologies UK Ltd
Current assignee: NEC Technologies UK Ltd
Priority date: 2006-03-15
Filing date: 2006-03-15
Publication date: 2007-02-21
Anticipated expiration: 2026-03-15
Also published as: GB2429346B; GB0605202D0

Abstract

A method of adjusting the amplitude of an audio signal (302) is described. The method comprises a user (230) selecting or pre-determining user selected minimum (318) and maximum (314) output audio signal amplitudes, through, for example, either preset profile selection or an in-situ calibration process; and a processor processing or transforming the input audio signal (302) into an output audio signal such that the output audio signal amplitude is within minimum (318) and maximum (314) output audio signal amplitudes derived from the user selected minimum and maximum output audio signal amplitudes. The processing occurs unless the output audio signal conflicts with another pre-determined criterion like, for instance, dynamic range preservation. A corresponding device (100) is also described.

Description

A METHOD OF ADJUSTING THE AMPLITUDE OF AN AUDIO SIGNAL AND

AN AUDIO DEVICE

Background of the Invention

The present invention relates to a method of adjusting the amplitude of an audio signal and an audio device.

Electronic devices which produce sound such as, for example, DVD players, video cassette recorders, televisions, personal stereos, and in particular mobile phones, are used in different environments to produce sound in different frequency ranges, which need to be heard by a user in these very different environments. For example, a mobile phone could be used outside in a noisy street environment to listen to music. Alternatively, a mobile phone could be used in a quiet office environment to listen to speech. In both these environments, one would wish to hear the sound clearly without other people in the neighbouring environment being subjected excessively to the sound, so called noise pollution, particularly as noise pollution is an increasing factor of stress and general unease in today's society. Furthermore, in extreme cases, noise pollution and excessive volume from an audio device can result in hearing impairment. Some typical volume levels are set out in the table below: Noise Source dBA Jet Aircraft at 30m (take-off) 130 Heavy Road Traffic at Sm 90 Business Office 60 Library 30 Bedroom at Night 20 Threshold of Hearing 0 On the other hand, the same mobile phone or electronic device could be used by a person who has a hearing impairment and thus generally requires a higher volume for the sound they wish to listen to than a person who does not suffer from a hearing impairment.

Furthermore, users of audio equipment, and in particular mobile phones, have different preferences as to the volume of the sound they listen to because they prefer to hear audio at a particular high or low volume, which may depend on the type of music or speech being listened to, because they prefer to listen at different volumes depending on the environment they are listening in, for example, walking along a busy road or on a train, or because they have some type of hearing impairment, for example.

US patent No. US5907823 discloses an arrangement for adjusting the dynamic range of an audio signal in a mobile phone in transmission and reception depending on the audio signal level, audio noise level, ambient noise level and lower and upper threshold values.

US patent No. US6298247 discloses an apparatus for automatically controlling the volume level of an audio signal transmitted to or received by a mobile station and/or terminal within a telecommunications network, the apparatus includes a noise-measuring device. The volume level of the audio signal is adjusted dependent on the background noise level measurement. The gain of the desired signal and the gain of the undesired noise can be adjusted according to the user's preferences.

US patent No. US631 7613 discloses a method of achieving a desired received audio dynamic compression factor at higher background noise levels.

The received audio signal remains compressed for local background noise levels, measured using a microphone, above a predetermined threshold level.

Hearing aid devices are known which have a sound peak prevention function, which limits audio volume by "flattening" sound peaks.

Television sets are known which automatically lower the volume level of a television set by several dB when a commercial break in a broadcast causes the broadcast volume to increase by a few dB. This allows the TV set user to experience a stable volume level throughout the whole broadcast, irrespective of whether or not a commercial break causes a broadcasted sound level to change.

The Sony DVD player model reference DVP-NS355 features a "night mode" in which the volume of a movie or audio file being played is limited.

There is however a need for a generic method of adjusting the amplitude of an audio signal and for an audio device that can be easily adapted to operate at the volume requirements and preferences of different users in different operating environments.

Summary of the Invention

The invention is defined in the independent claims below to which reference should now be made. Advantageous features are set forth in the appendant claims.

A preferred embodiment of the invention is described in more detail below and takes the form of a method of adjusting the amplitude of an audio signal.

The method comprises a user selecting or pre-determining user selected minimum and maximum output audio signal amplitudes through, for example, either preset profile selection or an in-situ calibration process; and a processor processing or transforming the input audio signal into an output audio signal such that the output audio signal amplitude is within minimum and maximum output audio signal amplitudes derived from the user selected minimum and maximum output audio signal amplitudes. The processing occurs unless the output audio signal conflicts with another pre-determined criterion like, for instance, dynamic range preservation. A corresponding device is also described.

Brief Description of the Drawings

The invention will be described in more detail by way of example with reference to the accompanying drawings, in which: Figure 1 is a block schematic diagram showing an audio device embodying the present invention; Figure 2 is a flow diagram showing the operation of the audio device of Figure 1; Figure 3 is a schematic diagram showing the operation of the device of Figure 1; Figure 4 is a flow diagram showing the calibration method used by the device of Figure 1; Figure 5 is a flow diagram showing the "preset profile" method used by the device of Figure 1; Figure 6 is a flow diagram showing the "dynamic preserving" method used by the device of Figure 1; Figure 7 is a schematic diagram illustrating the "dynamic preserving" method of Figure 6; Figure 8 is a schematic diagram illustrating the sample period used by the "dynamic preserving" method of Figure 6; Figure 9 is a schematic diagram illustrating an alternative sample period used by the "dynamic preserving" method of Figure 6; Figures 10, 12, 14 and 16 are tables that illustrate the "dynamic preserving" method of Figure 6; Figures 11, 13 and 15 are schematic diagrams illustrating the "dynamic preserving" method of Figure 6; Figure 17 is a table illustrating the sample periods used by the "dynamic preserving" method of Figure 6; Figure 18 is a flow diagram showing the "peak clipping" method used by the device of Figure 1; Figures 19, 21, and 23 are schematic diagrams illustrating the "peak clipping" method algorithm of Figure 18; and Figures 20, 22 and 24 are tables illustrating an example of the "peak clipping" of Figure 18.

Detailed Description of the Preferred Embodiment

A preferred audio device 100 and method will now be described with reference to Figures 1, 2 and 3. The device 100 comprises a user input unit 102 having a selector 103 for selecting the type of calibrating sound sample a user plans to use as most representative of the sound that he will be listening to in the calibration method and a plurality of push buttons 104 to select the desired maximum and minimum volume level or output minimum and maximum amplitude of the audio device 100, and to choose between preset profiles and calibration methods, or to save a custom profile a into the preset profiles. The preset profiles and calibration methods are described below. In this example, the selector 103 is a sliding multi-position switch that has a plurality of signs 101 which indicate the type of calibrating sound sample the user plans to listen to (for the calibration method). Note that the selector 103 is irrelevant if the user chooses to select a preset profile instead of calibrating his own.

The user input unit 102 is connected to three memory devices: an audio sample memory device 105, a volume level memory device 106, and preset profile memory device 107. Audio sample memory device 105 stores audio samples for use in the calibration process described below, for setting the maximum and minimum volume level of the audio device 100. The audio sample memory device 105 is connected to audio adjusting device 108, so as for itto be able to play the audio samples stored in the memory device 105. Volume level memory device 106 stores the user-selected desired maximum and minimum volume level and is connected to the audio adjusting device 108 so as to be able to provide these audio level parameters to the audio adjusting device 108 as input parameters for its audio processing. Preset profile memory device 107 stores a minimum and maximum volume level associated for each sound type or preset profile that a user may wish to listen to and is also further connected to the volume level memory device 106 so as to set its minimum and maximum volume values to the preset levels stored in preset profile memory device 107.

Additionally, the user input unit 102 allows the device 100 to store the volume level memory parameters from the volume level memory 106 as some preset minimum and maximum volume levels into some suitably useradjustable preset profiles, inside the preset profile memory, thereby allowing the user to recall this custom profile at a later time, without having to go through the calibration process (described below) all over again. Speech and music are illustrated as sound types in Figure 1.

The audio signal processing device 108 is also connected to an audio producing device 110 (which may be a mobile phone, a CD player, a DVD player or a personal computer, for example) and the audio output device 112 (loudspeaker). The audio signal processing device 108 can process the audio signal produced from the audio producing device 110 so that the audio produced is within the maximum and minimum volume levels set in the volume level memory device 106.

In use, a user 230 first specifies his audio requirements 202, 232, (see Figures 2 and 3). There are a number of ways in which this can be done, for example, using a calibration process (illustrated in Figure 4) or using preset profiles (illustrated in Figure 5) and both described below. The audio signal is then processed using a method such as the peak clipping method or the dynamic preserving method described below (step 204, Figure 2). Audio is then emitted based on the processed audio signal (step 206, Figure 2).

Each of these stages is described below.

Calibration Process The user 230 can specify his audio requirements using the calibration process. The user 230 first initiates the process 250 (Figure 4) by pushing a push button 104a. The user 230 then selects the type of sound sample 252 he intends to listen to by moving selector 103 so that it is aligned with the sign 101 representing the particular type of sound he wishes to listen to, for example a complex audio sample such as speech or music, or alternatively, the selector 103 can be aligned with a sign 101 of a simple sample, such as a pulse or tone at several different frequencies within the audio spectrum as a representative sound, so as to fully cover the user audible band. Pushing push button 104a again starts the playing of an audio sample 254, stored in the audio sample memory 105 representative of the sound indicated by the selector 103 to be played through audio output device 112. The sound has a crescendo rising volume that starts just below the normal sound level where a person can hear and rises slowly to a loud level. This audio sample is played repeatedly. A user 230 states when he can hear the audio sample 256 by pressing push button 1 04a again and as illustrated at 258 of Figure 4. The volume or loudness level at this time is stored in the volume level memory device 106 in the memory location for the minimum sound level 106a (step 260). Meanwhile, the audio sample continues to be played 262. A user 230 states when the audio sample has just become too loud 264 by pressing push button 1 04a yet again (step 266). The loudness or volume level at this time is stored in the volume level memory 106 in the memory location for the maximum sound level 1 06b (step 268). Once the minimum and maximum volume level are selected the audio sample is stopped 270.

Note that the user can then use another sound type by moving the selector 103 onto another sound type indicated by a sign 101, and that the results of minimum and maximum levels will be combined with the previous results onto the volume level memory 106. The previous volume level memory 106 setting can be erased so as to only account for the new calibration results by pressing in short succession the buttons 104a, 104b, 104a.

New volume level memory 106 settings are combined with the new volume level results by taking the lowest value of the maximum sound levels (provided it does not become lower than the minimum sound level value) and the highest value of the minimum sound levels (provided it does not become higher than the maximum sound level value) and storing these results in the volume level memory 106.

Preset Profiles The user 230 can specify his audio requirements using "preset profiles".

To use a preset profile, the user 230 first initiates the process by pushing push button 104b (step 270, Figure 5). The user 230 then pushes appropriate buttons 104 in order to select either a user defined preset, or a preset that is governed by three user-selected parameters: the state of the user hearing (for example, perfect, early impairment, late impairment, using hearing aid), the noise environment (for example, plane, train, office, home), and the sound type that the user plans on hearing (for example, speech, movie, rock music, classical music, games sounds). This causes the minimum and maximum volume levels stored in the preset profile memory device 107 to be stored in the memory location for the minimum volume level 1 06a and maximum volume level 1 06b of the volume level memory device 106 (step 274). The selector 103 may have preset profiles that reflect the particular sound environment where the user intends to listen to audio such as on an underground train, in a plane, in an office, at home, in an open space, in a windy place, in a human-voice-noisy place, or using closed or open earphones. It may reflect different audio spectra such as those required by a hearing aid user, a user with typical early hearing impairments, or with perfect hearing. There may be a greater selection of possible audio types rather than simply speech and music, for example, movie sound, sound from a games console or even particular types of music such as rock, classical or techno-music.

These types of settings influence the maximum volume level and the minimum volume level as well as the frequency band that needs specific volume boosting or capping.

The preset profiles are particularly useful when the user 230 does not want to spend time to go through a calibrating process and has a rough idea of his hearing profile, his environment and the type of audio he will be listening to.

Once the minimum and maximum volume levels have been set, the audio played on the audio producing device 110 is adjusted or re-formatted by the audio adjusting device or audio signal processing device 108 (step 204 shown in Figure 2) using one of a number of suitable methods (the "true volume" method 234 of Figure 3). Two example methods are described below. They are the "dynamic preserving" method and the "peak clipping" method. The dynamic preserving method preserves dynamic range while ensuring that the sound level in the sample does not reach above the "optimum volume comfort level" of (maximum volume level - a margin level). The dynamic preserving method increases the volume of the audio having a volume level below the minimum volume set by the user, or decreases the volume if the audio would rise above the maximum audio level set by the user. The priority is set on the maximum level which can not be surpassed by the audio level in the sample, and as the dynamic range is to be maintained, that could mean that in places the audio volume may be below the minimum threshold set by the user. Note however that the sound dynamic range (maximum volume level minimum volume level) of the audio produced by the audio producing device 110 is not changed by the audio output device 112 in this method.

The peak clipping method reduces the dynamic range of the input audio signal to fit in the specified dynamic range of [(maximum volume level - a margin associated with the maximum volume level)) - (minimum volume level + a margin associated with the minimum volume level))] and adjusts the sound levels of the input audio signal accordingly.

Dynamic Preserving Method There are three phases to the operation of the dynamic preserving method. These are illustrated in Figures 6 to 15.

In the first phase 300, the audio signal 302 from the audio producing device 110 is sampled over a time period 304 or a specific audio sample period (step 400, Figure 6) and stored in a memory (not shown). The sample time period 304 is dependent on the application.

As illustrated in Figure 8, for sound playback from a pre-recorded audio source such as a CD or DVD the full audio clip! song is analysed giving a sample time period 236, 304 of several minutes to several hours. Alternatively, for these applications, a smaller sample time period 236, 304 can be used (for example, one minute or the length of a chapter of a chapterised DVD audio file). Shorter sample time periods 236, 304 result in shorter processing time by the signal processing device 108. If then sample time periods are too short, sound may seem hacked to the user by possible brutal volume changes in between sampling periods. This can be tackled by introducing an additional constraint of a maximum allowed change in volume between two consecutive sampling periods.

As illustrated in Figure 9, for applications that do not playback prerecorded audio, but use "real-time" audio such as a telephone or a hearing aid, the sample period 238,304 needs to be quite short, in the range of milliseconds or seconds, so as not to generate a noticeable delay in the sound retransmission.

The sample period could be 10 to 2Oms because this is the time period used in a GSM phone for audio sampling, and as such the sound distortion should be limited and acceptable. These short sample times 238, 304 prevent excessive delay of the audio before it is heard by the user 230.

The table of Figure 15 illustrates in more detail the different sample or analyser periods 236, 304 that are used.

For pre-recorded audio the period of audio analysis may be typically from 1 second to up to 20 seconds, thereby analysing audio samples whose realtime' duration should be in between 5 seconds to up to 4 hours. For a better user experience, the period of audio analysis should be based on the typical lag times of the sound producing devices: up to 20 seconds for hi-fi, audio players and for video players (the time should be approximately the time to start the player), less than 5 seconds for broadcast receivers such as television and radio (the time is less than the time the broadcaster allows for editing and camera choice), and less than 1 second for simple music and tones, such as from a personal computer, a personal digital assistant, a mobile phone or other multimedia equipment such as for boot tones or ring tones (this short time period is used only for sounds that are not too time critical).

For "real-time" applications, where a significant delay of analysis could not be tolerated, the analysis time needs to be much shorter in the range of 2 to 5ms, depending on the application. The duration of the sample should also be in the range of 20 to lOOms as otherwise it provokes a direct delay for sound rendering.

This should be compared with the specifications for the round-trip delay of sound in a GSM mobile phone which is of about 1 5Oms: 2Oms sampling time is possible (and is actually the period of sampling time for audio processing in GSM mobile phones. Please refer to the second part of Figure 15, which gives typical time requirements for both the audio sample duration and the duration for processing (analysing) this sample, for most real time applications. Note that out of all three processing phases for the audio sample in this algorithm, it is expected that phase 1 (sampling/analysing) would be the most delay inducing. That is why analysis period is used as well as processing period to specify the overall delay induced in the sound rendering by the implementation of the algorithm.

In the second phase 316, the following pre-determined criteria are followed. The audio sample 302 is analysed to ensure that the audio volume or volume level 312 is below the maximum volume level 314 stored in the volume level memory device 106 as illustrated in step 402. If at any time over the sample or analysis period 304 the audio volume 312 exceeds the maximum volume level 314, then digital signal processing is performed to decrease the volume level 312 of the whole sample period 304, so that it does not exceed a level of (maximum volume level - margin) wherein the margin is typically 3dBA (margin could be optimised by the user 320) as illustrated at step 404. The decrease in volume is made uniformly over the entire sample period 304.

The reduction of the volume level over the entire signal 302 in the sample period 304, will then be reduced according to the following equation: Mmax + mmax = Tmax (1) where Mmax = the maximum sound level above the maximum volume level; mmax = the margin; and Tmax = volume decrease applied over the whole sample period.

If the volume level 312 has been decreased, no further changes will be made to the audio signal 302 as illustrated by 406 of Figure 6. If the volume level 312 has not been decreased, then the third phase 310 of the processing is carried out.

In the third phase 310, the audio signal 302 over the sample period 304 is analysed to check whether, at any time in the sample period 304, the audio signal 302 drops below the minimum volume level 318 (step 408). If the audio signal does drop below the minimum volume level at any time point, then an increase in the volume level 312 of the audio signal 302 over the whole sample period 304 will be considered according to the following equation: Mmin + mmn = Tmin (2) where Mmn = the maximum sound level below the minimum volume level; mmn = the margin; and TmIn = volume increase applied over the whole sample period.

If the proposed increase by Tmn would result in any portion of the audio signal 302 in the sample period 304 being above the maximum volume level 314 (step 410) then the audio signal level 312 is increased (step 412), but only so that the maximum sound level over the whole sample period is equal to the maximum volume level - margin in which case Tmin = maximum volume level - margin - maximum second level. If the proposed increase by Tmn would not result in any portion of the audio signal 302 in the sample period 304 being above the maximum volume level 314 then the audio signal level 312 is increased over the entire sample period 304 (step 414).

In the case of a uniform and very low-level sound sample, the third phase 310 may bring the audio signal level or volume level 302 well above the minimum volume level 318 because this volume level 302 will be (maximum audio level - a margin). The dynamic preserving algorithm uses this high level as the default optimum comfort level for the user 230.

Referring to Figures 7 and 10 to 16 four examples A, B, C and D of the second and third phases are illustrated and described below.

In example A of Figures 7 and 10, as shown in the first phase 310, the sample analysis period 304 of the audio signal 302 has two peaks 306 and 307 that exceed the maximum volume level 314 of 67dBA, one peak 306 by 1. 7dBA and the other peak 307 by 8.3dBA, and for the rest of the sample period 304 the audio signal 302 is at a level below the maximum volume level 314. Then Mmax would be 8.3dBA. In this case the margin mmax is selected as 3dB. Hence, using equation 1, Tmax = 8.3dBA + 3dB = 11.3dBA. Hence, in phase two, there will still be two peaks 306 and 307, but they will be at a level of (maximum volume level - 11.3dBA) = 57.4dBA and (maximum volume level - 11.3dBA) = 64dBA and for the rest of the time the audio signal 302 will be below these levels. In this example, as the volume level is changed in the second phase 316, the method does not continue to the third phase 310.

In example B of Figures 11 and 12, as shown in the first phase 310, the sample analysis period 304 of the audio signal 302 has one peak 307 that exceeds the maximum volume level 314 of 67dBA by 4.3dBA, and for the rest of the sample period 304 the audio signal 302 is at a level below the maximum volume level 314. Then Mmax would be 4.3dBA. In this case the margin mmax 15 selected as 3dB. Hence, using equation 1, Tmax = 4.3dBA + 3dB = 7.3dBA.

Hence, in phase two, there will still be one peak 307, but it will be at a level of (maximum volume level - 7.3dBA) = 64dBA and for the rest of the time the audio signal 302 will be below this level. In this example, as the volume level is changed in the second phase 316, the method does not continue to the third phase 310. Note that the lower peak 320, which was already below the minimum audio level 318 before the algorithm implementation, is now even further below this minimum audio level 318 due to the presence in the same audio sample 302 of an upper peak 307 that was above the maximum audio level 314, on which the priority went for adjustment.

In example C of Figures 13 and 14, as shown in the first phase 300, the sample analysis period 304 of the audio signal 302 does not have any peaks that exceed the maximum volume level 314. Hence, no change is made to the audio signal 302 in the second phase 316. However, the audio signal 302 has two troughs 320 and 322 that extend below the minimum volume level 318, one trough 320 by 2.9dBA and the other trough 322 by 7. 5dBA. Therefore, Mmin would be 7.5dBA. In this case the margin mmn is selected as 3dB. Hence, using equation 2, Tmn = 7.5dBA + 3dB 10.5dBA. However, increasing the volume level over the entire sample analysis period 304 by 10.5dBA would result in, at some time point, the volume level exceeding the maximum volume level 308.

Hence, the volume level is only increased such that at its highest point, the volume level is equal to the maximum volume level - margin, which in this case is 67dBA - 3dBA = 64dBA. In this example, the highest volume level in the sample analysis period 308 is 54.6dBA, hence the maximum increase in volume level in the sample analysis period is maximum volume level - margin - maximum sound level = 67dBA - 3dB - 54.6dBA = 9.4dB. Hence, there will still be two troughs 320 and 322, but they will be at a level of (minimum volume level + 6. 5dBA) and (minimum volume level + 1.9dBA).

In example D of Figures 15 and 16, the audio signal 302 is above the minimum volume level 318 and below the maximum volume level 314 throughout the audio sample period 304. Hence, the audio signal 302 is not changed in either the second phase 316 or the third phase 310.

In an alternative embodiment, if an increase in the audio signal 302 in the third phase would result in the maximum sound level exceeding the maximum volume level - margin (max), then no change is made at all to the audio signal 302.

Peak Clipping Method As mentioned above, as an alternative to the dynamic preserving method the peak clipping method can be used. This method is illustrated in Figures 18 to 24 and comprises three phases or steps.

In the first phase or step 500, the audio signal 502 is sampled over a specified sample period 504 or specific audio sample period and the audio signal 502 over this time period 504 is stored in a memory (step 600, Figure 8). The sample period 504 is dependent on the particular application and is of the same range of periods as set out for the dynamic preserving method above. The average sound level (AVG) is then calculated for this whole sample period 504 and stored in a memory (step 602). AVG iscalculated by, firstly, adding together the audio volume level at each sample point and then, secondly, dividing the result by the total number of sample points.

In the second phase 506, the following pre-determined criteria are taken into account. The audio signal 502 at every sample point is checked to see if it lies outside the user-specified dynamic range (specified using the calibration method or preset profiles described above), which is (user-specified maximum volume level - the user-specified minimum level), when the dynamic range is centred on AVG (step 606). If the audio signal 502 at a particular sample point has a value exceeding the user-specified dynamic range including margins, this sample's volume level is levelled, cut-off or clipped to fall within the user-specified dynamic range including margins (step 608).

Typically margin (max) and margin (mm) are both set to 3dB, but they could be specified or optimised by the user.

In the third phase 508, the overall level of the audio sample 502 is adjusted or processed to fit user specifications. That is to say, to fit between user-specified maximum volume level - margin (max) and userspecified minimum volume level + margin (mm). This is done by a simple increase or decrease in overall volume of the audio sample so that its AVG is aligned with the user-specified average (= (minimum level + maximum level) I 2).

In one arrangement, the audio signal 502 is adjusted in volume level over all sample points so that the maximum value of all the sample points over the entire sample period 504 is equal to (maximum volume level - margin (max)) (step 610). This provides a higher average volume level, which should enhance the audibility of low volume parts.

Referring to Figures 19 to 24, three examples E, F and G of the first, second and third phases are illustrated and these are described below.

In example E of Figures 19 and 20, margin (mm) is set at 3dB, margin (max) is set at 3dB, the maximum volume level is set at 65dBA, and the minimum volume level is set at 35dBA. In this example, in the sample period 504 there are two peaks 510 and 512 in the audio signal 502 that exceed the maximum volume or audio level 514. One peak 510 has a volume level 514 of 72.6dBA, the other peak 512 has a volume level of 79.2dBA. In the rest of the sample period 504, the audio signal 502 is at a volume level below the maximum volume level and above the minimum volume level including margins. The audio signal is sampled and stored (step 600). The average volume level AVG is then calculated (step 602). In this example, AVG = 54. 8dBA. The user-specified dynamic range is calculated as user-specified maximum volume level - user- specified minimum level = 65dBA - 35dBA = 30dB. The maximum allowable volume level is therefore the average volume level + half the dynamic range = (54.8 + (30/2)) = 69.8dBA and the minimum allowable volume level is the average volume level - half the dynamic range = (54.8 - (30/2)) = 39.8dBA. The stored signal is then checked to see whether the volume levels fall within these values including margins of, in this case, 3dB (step 606). That is to say, whether they fall within the range of 66.8dBA to 42 8dBA If not, the volume levels are clipped to these values. In this case, the two peaks 510 and 512 fall above the maximum value and they are clipped to the maximum value of 66.8dBA (step 608). In the third phase, the level of the clipped audio signal 502 is adjusted in volume at all sample points so as to match the user specified audio average level (= (Max + Mm) I 2) reduced in volume by the user specified average audio level - audio sample 502 average audio level = 4.8dBA.

In example F of Figures 21 and 22, margin (mm) is set at 3dB. Margin (max) is set to 3dB too. The maximum volume level is set at 65dBA and the minimum volume level is set at 35dBA. These are the same values as example E. In exampleF, there are two troughs 516 and 518 in the audio signal 502 that go below the minimum volume or audio level 514. In the rest of the stored sample period 504 (step 600), the audio signal 502 is at a volume level below the maximum volume level and above the minimum volume level including margins.

In this example, the average audio level calculated in phase 1 at step 602 is AVG = 45.2dBA. The user-specified dynamic range is calculated, as before, as user-specified maximum volume level - user-specified minimum volume level = 65dBA - 35dBA = 3OdBA. The maximum allowable volume level is therefore the average volume level + half the dynamic range = (45.2 + 30)/2 = 60. 2dBA and the minimum allowable volume level is the average volume level - half the dynamic range = (45.2 - (30/2)) = 30.2dBA. The stored signal is checked to see whether the volume levels fall within these values including margins of, in this case, 3dB (step 606). That is to say, whether they fall within the range of 33.2dBA to 57.2dBA. If not, the volume levels are clipped to these values. In this case, the two troughs 516 and 518 fall below the minimum allowable volume level of 33.2dBA. They are, therefore, clipped to this minimum volume level of 33.2dBA (step 608).

In the third phase 508, the level of the clipped audio signal 502 is adjusted in volume at all sample points so as to match the user specified average audio level (= (Max + Mm) / 2) increased in volume by user specified average audio level - audio sample 502 average audio level = 4. 8dBA.

In example G of Figures 23 and 24, the same maximum and minimum volume level, and margins, are set as examples E and F. In example G, the audio signal 502 extends beyond the minimum and maximum volume level (including margins) at a number of sample points.

In this example the average audio level calculated at phase 1 at step 602 is AVG = 38dBA. The user-specified dynamic range is calculated, as before, as user-specified maximum volume level - user-specified minimum volume level = 65dBA - 35dBA = 30dB. The maximum allowable volume level is therefore the average volume level + half the dynamic range = 38 + (30/2) = 53dBA and the minimum allowable volume level is the average volume level - half the dynamic range = 38-(30/2) = 23dBA.

The stored signal is checked to see whether the volume levels fall within these values including margins of, in this case, 3dB (step 606). That is to say, whether they fall within the range of 26dBA to 5OdBA. If not, the volume levels are clipped at these values. In the example points illustrated in Figure 22, at sample point no. 71 the sound level is 23. 7dBA and falls below the minimum allowable level and hence the volume level at this sample point is clipped to 26dBA. At sample points nos. 72 and 192 the sound level is 60.9dBA and 71.3dBA respectively, which are above the maximum allowable level. Hence, the volume level at this sample point is clipped to the maximum allowable level of 5OdBA.

In the third phase 508, the level of the clipped audio signal 502 is adjusted in volume at all sample points so as to match the user specified a average audio level (= (Max + Mm) / 2) increased in volume by user specified average audio level - audio sample 502 average audio level = ((65+35)12) - 38 = 50 - 38 = 12dB.

The audio signal processing described above can be performed in a variety of equipment types. The signal processing can be carried out within audio reproducing equipment such as a mobile phone, personal computer, personal digital assistant, hi-fi, television, personal stereo, or MP3, mini disc, tape, compact disc, DVD, advanced optical disc (AOD) or video player; within the sound emitting device or peripheral such as head phones, ear phones, speakers, audio amplifier or hearing aid device (as an add-on to a standard hearing aid device or as part of an integral device) by integrating the necessary digital audio processing into the software or hardware of the equipment. The signal processing can be carried out in an add-on unit to the above-mentioned devices, which means that existing equipment can be easily modified to include this extra signal processing functionality. The digital audio processing device could be fitted at the sound output end of the equipment.

Typical sampling times and processing times for different applications are as follows.

20 milliseconds sampling time (or sample period) and less than 2 milliseconds processing time (or processing included delay) (so overall delay of less than 22 milliseconds) for the mobile phone application. 5 seconds sampling time and less than 5 seconds processing time (so overall delay of less than 10 seconds) for broadcast receiver applications, like but not restricted to TV, radio, mobile TV. 30 seconds sampling time and less than 1 second processing time (so overall delay of less than 1 second as the full tune is readily available) for the production of simple non-interactive music or tones by PC (personal computer), PDA (personal digital assistant), mobilephones, or other multimedia equipment.

The duration of a stored full file or media (movie, or clip, or song, or album) up to 4 hours sampling time and less than 20 seconds processing time for the production of simple non-interactive music or tones by PC, PDA, mobile phones, or other multimedia equipment. Various modifications can be made to the preset profiles system.

Preset profiles can be provided that reflect both the type of sound and the listening environment. For example, "speech in a noisy background" is a listening mode where the speech frequencies are particularly increased in volume, with a quite high maximum and minimum volume level; and "movie at night" is a listening mode where the minimum level is medium low and the maximum level is medium. Other listening modes can include "instrumental music in a noisy

background".

Preset profiles that can be selected by the user include minimum and maximum audio signal amplitude according to three parameters: user hearing preference, noise environment, and audio characteristics, or a previously set-up and saved profile setting. The user hearing preference could be related to user hearing impairment level, wearing a hearing aid, or sound level preferences like loud or quiet. The noise environment could be a high noise environment, like a plane or a train, or a low noise environment like at home. The audio characteristics are related to the type of sound that the user plans on listening to speech which has a specific bandwidth used only, movie which combines many kinds of sounds, rock music, classical music, jazz, or synthetic sounds like games. The previously saved setting could be the result of a long and difficult calibration session with several additional tweaks, that the user has perfected for an optimum user experience in specific conditions.

The preset profiles system can include a "test" mode. In the "test" mode, selected by the user, the user is played a sample of speech or music or a simple sample such as a tone or pulse according to the requirements of the selected preset profile. In this way, the user can make sure that the sound setting provided by the selected preset profile is adequate.

Embodiments of the audio device 100 and method can also allow a user to record his preferred settings in preset such as "user setting 1" and "user setting 2". The user can either edit an exiting preset profile using a menu system, for example. This menu system can allow a user to be given options to edit a user profile by, for example, being instructed to "turn the volume control until the played sample is at the maximum volume level you want to hear" or by using the calibration process described above.

Preset profiles can be generated by the equipment manufacturer in order to account for a typical user of the particular audio device 100. For example, a set-top box including preset profiles can have lower volume levels than a mobile multimedia player because the latter is more likely to be used in a noisy environment.

Different preset profile names can be used for different audio devices 100 and their particular use.

The equipment manufacturer can "fine-tune" the preset profiles according to a specific user category that the equipment will be used by. Alternatively, the preset profiles can be "fine-tuned" at the point-ofsale to meet particular customer or user requirements. User categories can include, for example, users who like loud audio or quiet audio or those who prefer genuine sound above understandable voices and others who prefer to hear and understand and would accept distorted sounds in order to do this.

The device and method described herein can be implemented on an audio reproducing device, such as but not restricted to a personal computer, personal digital assistant, hi-fi, television, personal stereo, or MP3 player, mini disc player, tape player, compact disc player, DVD, player, Blu-ray disc (RTM) player, advanced optical disc (AOD) player, HD-DVD (high definition DVD) player or video player or a sound emitting device or peripheral, such as but not restricted to, head phones, ear phones, speakers, audio amplifier or hearing aid device.

Embodiments of the present invention have been described with particular reference to the examples illustrated. However, it will be appreciated that variations and modifications may be made to the examples described within the scope of the present invention.

Claims

1. A method of adjusting the amplitude of an audio signal, the method comprising: a user selecting user selected minimum and maximum output audio signal amplitudes; and a processor processing an input audio signal into an output audio signal such that the output audio signal amplitude is within minimum and maximum output audio signal amplitudes derived from the user selected minimum and maximum output audio signal amplitudes, unless the output audio signal conflicts with a pre-determined criterion.

2. A method according to claim 1, wherein if the amplitude of the output audio signal in a sample period is above the output maximum amplitude at any time, processing the input audio signal by decreasing the entire input audio signal level by the same amount such that the amplitude of the entire output audio signal in the sample period is equal to or below the output maximum amplitude.

3. A method according to claim 2, wherein the entire output audio signal is further decreased in level below the output maximum amplitude by a predefined margin.

4. A method according to claim 3, further comprising the step of a user defining the pre-defined margin.

5. A method according to claim 3, wherein the pre-defined margin is substantially 3dB.

6. A method according to claim 1, wherein if the amplitude of the input audio signal is below the output minimum amplitude at any time in a sample period, processing the input audio signal by increasing the entire input audio signal level in the sample period by the same amount such that the amplitude of the entire output audio signal is equal to or above the output minimum amplitude.

7. A method according to claim 6, wherein the entire output audio signal level is further increased above the output minimum amplitude by a predefined margin.

8. A method according to claim 7, further comprising the step of a user defining the pre-defined margin.

9. A method according to claim 7, wherein the pre-defined margin is substantially 3dB.

10. A method according to any of claims 6 to 9, comprising the step of increasing the entire input audio signal level only if the increase does not result in any part of the output audio signal amplitude becoming greater than the maximum output audio signal amplitude.

11. A method according to claim 1 comprising: calculating the average signal level value of the entire input audio signal in a sample period calculating, at a plurality of time points of the input audio signal, the difference between the amplitude of the input audio signal at a time point and the average signal level value; and if the difference exceeds a pre-determined proportion of [(user-specified output maximum amplitude - a first pre-determined margin) - (user- specified output minimum amplitude a second pre-determined margin)], giving the output audio signal the value of the average signal value plus (if positive difference) or minus (if negative difference) the pre- determined proportion of [(user-specified output maximum amplitude - a first pre- defined margin) - (user-specified output minimum amplitude - a second pre- defined margin)]; then adjusting the input audio signal uniformly for substantially the whole sample period so that the average signal level value is brought to a level substantially equal to the pre-determined proportion of [(user-specified output maximum amplitude - the first pre-determined margin) + (user- specified output minimum amplitude - the second pre-determined margin)]..

12. A method according to any of claims 1 to 11, wherein the sample period of the input audio signal and processing-induced delay is adapted to the application where it is used.

13. A method according to claim 12, wherein the sample period is between substantially 20 milliseconds and substantially 4 hours.

14. A method according to claim 13, wherein the sample period is between substantially 5 seconds and substantially 30 seconds.

15. A method according to claim 13, wherein the sample period is between substantially 10 milliseconds and substantially 20 milliseconds.

16. A method according to any of claims 12 to 15, wherein the processinginduced delay is between substantially 2 milliseconds and substantially 20 seconds.

17. A method according to any of claims 12 to 16, wherein the processinginduced delay is between substantially 1 second and substantially 5 seconds.

18. A method according to claim 12, wherein the sample period is substantially 20 milliseconds and the processing-induced delay is less than substantially 2 milliseconds.

19. A method according to claim 12, wherein the sample period is substantially 5 seconds and the processing-induced delay is less than substantially 5 seconds.

20. A method according to claim 12, wherein the sample period is substantially 30 seconds and the processing-induced delay is less than substantially 1 second.

21. A method according to claim 12, wherein the sample period is substantially 4 hours and the processing-induced delay is less than substantially 20 seconds

22. A method according to any of claims 11 to 21, comprising the step of a user selecting at least one of the first and second pre-defined margins.

23. A method according to any of claims 11 to 21, wherein the first predefined margin is substantially 3dB.

24. A method according to any of claims 11 to 21 and 23, wherein the second pre-defined margin is substantially 3dB.

25. A method according to any of claims 11 to 24, wherein the piedetermined proportion is substantially 50%.

26. A method according to any preceding claim, wherein the step of a user selecting minimum and maximum output audio signal amplitudes includes: playing an audio sample of changing volume level over time; and a user selecting at least one of the minimum and maximum audio signal amplitudes based on the played audio sample.

27. A method according to claim 26, wherein the audio sample comprises an audio sample of increasing volume over time.

28. A method according to claim 26 or 27, wherein the user selects the minimum audio signal amplitude when the user hears an audio sample at the minimum level the user requires for the minimum audio signal amplitude.

29. A method according to any of claims 26 to 28, wherein the user selects the maximum audio signal amplitude when the user hears an audio sample at the maximum level the user requires for the maximum audio signal amplitude.

30. A method according to any preceding claim, wherein the minimum and maximum output audio signal amplitudes are selected according to at least one parameter from: user hearing preference, noise environment and audio characteristics related to sounds the user intends listening to.

31. A method according to claim 30, wherein the minimum and maximum output audio signal amplitudes are selected according to the three parameters of user hearing preference, noise environment and audio characteristics.

32. A method according to claim 30 or 31, wherein the user hearing preference is related to user hearing impairment level or use of a hearing aid or sound level preferences.

33. A method according to any of claims 1 to 29, wherein the minimum and maximum output audio signal amplitudes are selected from stored settings.

34. A method according to claim 33, wherein the stored settings are the result of a calibration by the user.

35. An audio device for adjusting the amplitude of an audio signal, the device comprising: a processor, the processor having an input for an input audio signal and an output for an output audio signal, the output audio signal having an output maximum amplitude and an output minimum amplitude, wherein the processor processes the input audio signal into the output audio signal; and a user controlled selector for a user to select user selected output maximum and output minimum amplitudes; the processor comprises a first processor for processing the input audio signal such that the maximum amplitude of the output audio signal is less than or equal to an output maximum amplitude derived from the user selected output maximum amplitude; and a second processor for processing the input signal such that the minimum amplitude of the output audio signal is greater than or equal to the output minimum amplitude derived from the user selected minimum amplitude, unless the output audio signal conflicts with a pre-determined criterion

36. An audio device according to claim 35, wherein the first processor is arranged such that if the amplitude of the input audio signal in a sample period is above the output maximum amplitude at any time, the first processor decreases the entire input audio signal level in the sample period by the same amount such that the amplitude of the entire output audio signal is equal to or below the output maximum amplitude.

37. An audio device according to claim 36, wherein the first processor decreases the entire input audio signal such that the amplitude of the entire output audio signal is further decreased in level below the output maximum amplitude by a pre-defined margin.

38. An audio device according to claim 37, wherein the audio device further comprises a margin definer arranged such that the user defines the pre- defined margin.

39. An audio device according to claim 37, wherein the pre-defined margin is substantially 3dB.

40. An audio device according to claim 35, wherein the second processor is arranged such that if the amplitude of the input audio signal is below the output minimum amplitude at any time in a sample period, the second processor increases the entire input audio signal level in the sample period by the same amount such that the amplitude of the entire output audio signal is equal to or above the output minimum amplitude.

41. An audio device according to claim 40, wherein the second processor is arranged such that it increases the entire input audio signal level such that the amplitude of the entire output audio signal is above the output minimum amplitude by a pre-defined margin.

42. An audio device according to claim 41, wherein the audio device further comprises a margin definer arranged such that the user defines the pre- defined margin.

43. An audio device according to claim 41, wherein the pre-defined margin is substantially 3dB.

44. An audio device according to any of claims 40 to 43, wherein the second processor is arranged such that it increases the entire input audio signal level only if the increase does not result in any part of the output audio signal amplitude becoming greater than the maximum output audio signal amplitude.

45. An audio device according to claim 35, wherein, the processor calculates the average signal level value of the entire input audio signal, in a sample period then, at a plurality of time points of the input audio signal within a sample period, the processor calculates the difference between the amplitude of the input audio signal at a time point and the average signal level value, if the difference exceeds a predetermined proportion of [(user-specified output maximum amplitude - a first pre-defined margin) - (user-specified output minimum amplitude - a second pre-defined margin)] , then the output audio signal is given the value of the average signal value plus (if positive difference) or minus (if negative difference) the pre-determined proportion of the [(user-specified output maximum amplitude - a first pre- defined margin) (user-specified output minimum amplitude - a second pre- defined margin)], then the processor adjusts the input audio signal uniformly for substantially the whole sample period so that the average signal level value is brought to a level substantially equal to the pre- determined proportion of [(user-specified output maximum amplitude - the first predetermined margin) + (user-specified output minimum amplitude - the second pre-determined margin)].

46. An audio device according to any of claims 35 to 45, wherein the sample period of the input audio signal and the processing-induced delay is adapted to the application where it is used.

47. An audio device according to claim 46, wherein the sample period is between substantially 20 milliseconds and substantially 4 hours.

48. An audio device according to claim 47, wherein the sample period is between substantially 5 seconds and substantially 30 seconds.

49. An audio device according to claim 47, wherein the sample period is between substantially 10 milliseconds and substantially 20 milliseconds.

50. An audio device according to any of claims 46 to 49, wherein the processing-induced delay is between substantially 2 milliseconds and substantially 20 seconds.

51. An audio device according to any of claims 46 to 50, wherein the processing-induced delay is between substantially 1 second and substantially 5 seconds.

52. An audio device according to claim 46, wherein the sample period is substantially 20 milliseconds and the processing-induced delay is less than substantially 2 milliseconds.

53. An audio device according to claim 46, wherein the sample period is substantially 5 seconds and the processing-induced delay is less than substantially 5 seconds.

54. An audio device according to claim 46, wherein the sample period is substantially 30 seconds and the processing-induced delay is less than substantially I second.

55. An audio device according to claim 46, wherein the sample period is substantially 4 hours and the processing-induced delay is less than substantially 20 seconds.

56. An audio device according to any of claims 45 to 55, wherein the audio device further comprises a margin definer arranged such that the user defines at least one of the first pre-defined margin and second predefined margin.

57. An audio device according to any of claims 45 to 55, wherein the first pre- defined margin is substantially 3dB.

58. An audio device according to any of claims 45 to 55 and 57, wherein the second pre-defined margin is substantially 3dBA.

59. An audio device according to any of claims 45 to 58, wherein the predetermined proportion is substantially 50%.

An audio device according to any of claims 35 to 59, wherein the user controlled selector further comprises an audio player for playing an audio sample of changing volume level over time; wherein the user selects at least one of the minimum and maximum audio signal amplitudes based on the audio sample played on the audio player.

61. An audio device according to claim 60, wherein the audio player plays the audio sample at increasing volume over time.

62. An audio device according to claim 60 or 61, wherein the user controlled selector further comprises means for selecting the minimum audio signal amplitude when the user hears an audio sample at the minimum level the user requires for the minimum audio signal amplitude.

63. An audio device according to any of claims 60 to 62, wherein the user controlled selector further comprises means for selecting the maximum audio signal amplitude when the user hears an audio sample at the maximum level the user requires for the maximum audio signal amplitude.

64. An audio device according to any of claims 35 to 63, wherein the user controlled selector allows the user to select the output maximum and output minimum amplitudes according to at least one parameter from: user hearing preference, noise environment and audio characteristics related to sounds the user intends to listen to.

65. An audio device according to claim 64, wherein the user controlled selector allows the user to select the output maximum and output minimum amplitudes according to the three parameters of user hearing preference, noise environment and audio characteristics.

66. An audio device according to claim 64 or 65, wherein the user hearing preference is related to user hearing impairment level or use of a hearing aid or sound level preferences.

67. An audio device according to any of claims 35 to 63, wherein the user controlled selector allows the user to select output maximum and output minimum amplitudes from output maximum and output minimum amplitudes stored in a storage means.

68. An audio device according to claim 67, wherein the output maximum and output minimum amplitudes stored in the storage means are the result of a calibration by the user.

69. An audio device as substantially hereinbefore described with reference to and as illustrated by the accompanying drawings.

70. A mobile communications device comprising the audio device according to any of claims 35 to 68.

71. An audio reproducing device comprising the audio device according to any of claims 35 to 68.

72. A sound emitting device comprising the audio device according to any of claims 35 to 68.

73. A method of adjusting the amplitude of an audio signal as substantially hereinbefore described with reference to and as illustrated by the accompanying drawings.

74. A mobile communications device comprising an implementation of the method according to any of claims 1 to 34 and 73.

75. An audio reproducing device comprising an implementation of the method of claims ito 34 and 73.

76. A sound emitting device comprising an implementation of the method of claims 1 to 34 and 73.

AMENDMENTS TO THE CLAIMS HAVE BEEN FILED AS FOLLOWS

A,-I.-.: -.

J. I ILI IOU Ull I LO I..,IQII I I J, VVI II Ii I II i JI II l'...1 II 1C41 substantially 3dB.

6. A method according to any preceding claim, wherein if the amplitude of the input audio signal is below the output minimum amplitude at any time in a sample period, processing the input audio signal by increasing the entire input audio signal level in the sample period by the same amount such that the amplitude of the entire output audio signal is equal to or above the output minimum amplitude.

11. A method according to claim 1 comprising: calculating the average signal level value of the entire input audio signal in a sample period calculating, at a plurality of time points of the input audio signal, the difference between the amplitude of the input audio signal at a time point and the average signal level value; and if the difference exceeds a pre-determined proportion of [(user-specified output maximum amplitude - a first pre-determined margin) - (user- specified output minimum amplitude a second pre-determined margin)], giving the output audio signal the value of the average signal value plus (if positive difference) or minus (if negative difference) the pre- determined proportion of [(user-specified output maximum amplitude - a first pre- S defined margin) (user-specified output minimum amplitude - a second S 25 pre-defined margin)]; S then adjusting the input audio signal uniformly for substantially the whole sample period so that the average signal level value is brought to a level S substantially equal to the pre- determined proportion of [(user-specified output maximum amplitude - the first pre-determiried margin) + (user- specified output minimum amplitude - the second pre-determined margin)].

16. A method according to any of claims 12 to 15, wherein the processinginduced delay is between substantially 2 milliseconds and substantially 20 r'rrdc 17. A method according to any of claims 12 to 16, wherein the processing- induced delay is between substantially 1 second and substantially 5 seconds.

21. A method according to claim 12, wherein the sample period is substantially 4 hours and the processing-induced delay is less than substantially 20 seconds.

25. A method according to any of claims 11 to 24, wherein the predetermined proportion is substantially 50%.

26. A method according to any preceding claim, wherein the step of a user selecting minimum and maximum output audio signal amplitudes includes: paying an audio sample of changing volume level over time; and a user selecting at least one of the minimum and maximum audio signal amplitudes based on the played audio sample.

32. A method according to claim 30 or 31, wherein the audio characteristics include frequency band.

33. A method according to claim 30, 31 or 32, wherein the user hearing preference is related to user hearing impairment level or use of a hearing aid or sound level preferences.

34. A method according to any of claims 1 to 29, wherein the minimum and maximum output audio signal amplitudes are selected from stored settings.

35. A method according to claim 34, wherein the stored settings are the result of a calibration by the user.

36. An audio device for adjusting the amplitude of an audio signal, the device comprising: a processor, the processor having an input for an input audio signal and an output for an output audio signal, the output audio signal having an output maximum amplitude and an output minimum amplitude, wherein the processor processes the input audio signal into the output audio V signal; and - V.- ____5_A__Il__I __I_ _L_ S __I__L. I__S__i _, ,4_, .1 d Uei (..,L) I ILl uiieu ll_,LL)I IL_Il UcI LU II_,L LAI UUI4JI..IL V maximum and output minimum amplitudes; V the processor comprises a first processor for processing the input audio signal such that the maximum amplitude of the output audio signal is less V than or equal to an output maximum amplitude derived from the user selected output maximum amplitude; and a second processor for processing the input signal such that the minimum amplitude of the output audio signal is greater than or equal to the output minimum amplitude derived from the user selected minimum amplitude, unless the output audio signal conflicts with a pre-determined criterion 37. An audio device according to claim 36, wherein the first processor is arranged such that if the amplitude of the input audio signal in a sample period is above the output maximum amplitude at any time, the first processor decreases the entire input audio signal level in the sample period by the same amount such that the amplitude of the entire output audio signal is equal to or below the output maximum amplitude.

38. An audio device according to claim 37, wherein the first processor decreases the entire input audio signal such that the amplitude of the entire output audio signal is further decreased in level below the output maximum amplitude by a pre-defined margin.

39. An audio device according to claim 38, wherein the audio device further comprises a margin definer arranged such that the user defines the pre- defined margin.

40. An audio device according to claim 38, wherein the pre-defined margin is substantially 3dB.

41. An audio device according to any of claims 36 to 40, wherein the second processor is arranged such that if the amplitude of the input audio signal is below the output minimum amplitude at any time in a sample period, the second processor increases the entire input audio signal level in the sample period by the same amount such that the amplitude of the entire output audio signal is equal to or above the output minimum amplitude.

42. An audio device according o ciaim 41, wh&en th second processor s arranged such that it increases the entire input audio signal level such that the amplitude of the entire output audio signal is above the output minimum amplitude by a pre-defined margin.

43. An audio device according to claim 42, wherein the audio device further comprises a margin definer arranged such that the user defines the pre- defined margin.

44. An audio device according to claim 42, wherein the pre-defined margin is substantially 3dB. 3(0

45. An audio device according to any of claims 41 to 44, wherein the second processor is arranged such that it increases the entire input audiosignal level only if the increase does not result in any part of the output audio signal amplitude becoming greater than the maximum output audio signal amplitude.

46. An audio device according to claim 36, wherein, the processor calculates the average signal level value of the entire input audio signal, in a sample period then, at a plurality of time points of the input audio signal within a sample period, the processor calculates the difference between the amplitude of the input audio signal at a time point and the average signal level value, if the difference exceeds a predetermined proportion of [(user-specified output maximum amplitude - a first pre-defined margin) - (user-specified output minimum amplitude - a second pre-defined margin)] , then the output audio signal is given the value of the average signal value plus (if positive difference) or minus (if negative difference) the pre-determined proportion of the [(user-specified output maximum amplitude - a first pre- defined margin) (user-specified output minimum amplitude - a second pre- defined margin)], then the processor adjusts the input audio signal uniformly for substantially the whole sample period so that the average signal level value is brought to a level substantially equal to the pre- determined proportion of [(user-specified output maximum amplitude - the first predetermined margin) + (user-specified output minimum amplitude - the second pre-deierrniried margin)].

S 47. An audio device according to any of claims 36 to 46, wherein the sample S period of the input audio signal and the processing-induced delay is adapted to the application where it is used.

48. An audio device according to claim 47, wherein the sample period is between substantially 20 milliseconds and substantially 4 hours.

49. An audio device according to claim 48, wherein the sample period is between substantially 5 seconds and substantially 30 seconds.

50. An audio device according to claim 48, wherein the sample period is between substantially 10 milliseconds and substantially 20 milliseconds.

51. An audio device according to any of claims 47 to 50, wherein the processing-induced delay is between substantially 2 milliseconds and substantially 20 seconds.

52. An audio device according to any of claims 47 to 51, wherein the processing-induced delay is between substantially 1 second and substantially 5 seconds.

53. An audio device according to claim 47, wherein the sample period is substantially 20 milliseconds and the processing-induced delay is less than substantially 2 milliseconds.

54. An audio device according to claim 47, wherein the sample period is substantially 5 seconds and the processing-induced delay is less than substantially 5 seconds.

55. An audio device according to claim 47, wherein the sample period is substantially 30 seconds and the processing-induced delay is less than substantially I second.

56. An audio device according to claim 47, wherein the sample period is substantially 4 hours and the processing-induced delay is less than substantially 20 seconds.

57. An audio device according to any of claims 46 to 56, wherein the audio device further comprises a margin definer arranged such that the user defines at least one of the first pre-defined margin and second predefined margin.

58. An audio device according to any of claims 46 to 56, wherein the first pre- defined margin is substantially 3dB.

59. An audio device according to any of claims 46 to 56 and 58, wherein the second pre-defined margin is substantially 3dBA.

60. An audio device according to any of claims 46 to 59, wherein the predetermined proportion is substantially 50%.

61 An audio device according to any of claims 36 to 60, wherein the user controlled selector further comprises an audio player for playing an audio sample of changing volume level over time; wherein the user selects at least one of the minimum and maximum audio signal amplitudes based on the audio sample played on the audio player.

62. An audio device according to claim 61, wherein the audio player plays the audio sample at increasing volume over time.

63. An audio device according to claim 61 or 62, wherein the user controlled selector further comprises means for selecting the minimum audio signal amplitude when the user hears an audio sample at the minimum level the user requires for the minimum audio signal amplitude.

64. An audio device according to any of claims 61 to 63, wherein the user controlled selector further comprises means for selecting the maximum audio signal amplitude when the user hears an audio sample at the maximum level the user requires for the maximum audio signal amplitude.

65. An audio device according to any of claims 36 to 64, wherein the user controlled selector allows the user to select the output maximum and output minimum amplitudes according to at least one parameter from: user hearing preference, noise environment and audio characteristics related to sounos the user intends to iisten to.

66. An audio device according to claim 65, wherein the user controlled selector allows the user to select the output maximum and output minimum amplitudes according to the three parameters of user hearing preference, noise environment and audio characteristics.

67. An audio device according to claim 65 or 66, wherein the audio characteristics include frequency band.

68. An audio device according to claim 65 or 66, wherein the user hearing preference is related to user hearing impairment level or use of a hearing aid or sound level preferences.

69. An audio device according to any of claims 36 to 64, wherein the user controlled selector allows the user to select output maximum and output minimum amplitudes from output maximum and output minimum amplitudes stored in a storage means.

70. An audio device according to claim 69, wherein the output maximum and output minimum amplitudes stored in the storage means are the result of a calibration by the user.

71. An audio device as substantially hereinbefore described with reference to and as illustrated by the accompanying drawings.

72. A mobile communications device comprising the audio device according to any of claims 36 to 70.

73. An audio reproducing device comprising the audio device according to any of claims 36 to 70.

74. A sound emitting device comprising the audio device according to any of claims 36 to 70.

75. A method of adjusting the amplitude of an audio signal as substantially hereinbefore described with reference to and as illustrated by the accompanying drawings.

76. A mobile communications device comprising an implementation of the method according to any of claims 1 to 35 and 75.

77. An audio reproducing device comprising an implementation of the method of claims ito 35 and 75.

78. A sound emitting device comprising an implementation of the method of claims 1 to 35 and 75.