Patent Abstract:
An apparatus for processing an input sound signal, the apparatus including: gain circuitry configured to control a gain based on a plurality of respective sub-signals of the input sound signal; and an amplification apparatus configured to adjust the amplification of all the plurality of amplitudes based on the common gain.

Full Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
     This application is a continuation application of U.S. Non-Provisional application Ser. No. 13/658,074, filed on Oct. 23, 2012, which claims priority to U.S. Patent Application No. 61/550,907, filed on Oct. 24, 2011 
    
    
     BACKGROUND 
     Field of the Invention 
     The present invention relates to post-filter processing of an input sound signal in order to adjust gain values. Such gain-control processing is applicable to hearing prostheses, telecommunications, and the like. 
     Related Art 
     Automatic Gain Control (AGC) systems are commonly used in audio processing systems (e.g., audio headsets, hearing prostheses, etc.) to cope with a large range in sound levels. In some systems, the audio signal is split into multiple frequency bands by a filter bank of discrete components or a transform (e.g., a Fast Fourier Transform). The gain of each band can then be controlled separately. This is referred to as a multi-band type of AGC. 
     A variety of hearing prostheses exist to assist people who suffer hearing loss. Some are entirely external devices, e.g., conventional hearing aids. Some hearing prostheses are implantable, and more particularly are examples of an active implantable medical device (AIMD). An AIMD is a medical device having one or more implantable components, the latter being defined as relying for its functioning upon a source of power other than the human body or gravity, such as an electrical energy source. Amongst hearing prostheses, an example of an AIMD is a cochlear implant system, which is used to treat sensorineural hearing loss by providing electrical energy directly to the recipient&#39;s auditory nerves via an electrode assembly implanted in the cochlea. Electrical stimulation signals are delivered directly to the auditory nerve via the electrode assembly, thereby inducing a hearing sensation (or percept) in the implant recipient. 
     When fitting a cochlear implant system to a recipient, the appropriate stimulation levels for each electrode must be determined. The lowest stimulation current that is perceptible is known as the threshold level or T level. The highest stimulation current that is comfortable is known as the maximum comfortable level or C level. The T and C levels vary between recipients, and also vary between electrodes in a single recipient. 
     The ratio between the C and T levels on an electrode is known as the electrical dynamic range, and is typically about 10 dB. This is much smaller than the dynamic range of sound levels in the environment, and hence the processing for a cochlear implant system generally incorporates some form of compression. 
     SUMMARY 
     In one aspect, an apparatus is provided. The apparatus comprises a frequency analysis unit configured to decompose a sound signal into a plurality of sub-signals each associated with a specific frequency band of the sound signal; amplitude detection circuits configured to produce provisional amplitudes for each of the sub-signals; gain circuitry configured to determine a common gain based on the provisional amplitudes; and an amplification module configured to generate a plurality of adjusted amplitudes by adjusting the amplification of all of the provisional amplitudes based on the common gain. 
     In another aspect, a method is provided. The method comprises generating provisional amplitude envelopes for a plurality of sub-signals that each comprise a frequency component of an input sound signal; generating, based on at least one gain rule, a common gain for application to each of the provisional amplitude envelopes; and applying the common gain to all of the provisional amplitude envelopes to produce a plurality of adjusted amplitude envelopes. 
     In another aspect, an apparatus is provided. The apparatus comprises a plurality of amplitude detectors configured to produce provisional amplitude envelopes for a plurality of sub-signals that each comprises a frequency component of an input sound signal; a level combiner configured to analyze the provisional amplitudes and generate a level signal; a gain rule module configured to generate a common gain based upon the level signal; and a bank of amplifiers configured to apply the common gain to the provisional amplitudes to generate adjusted amplitudes. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       Illustrative embodiments of the present invention are described herein in conjunction with the accompanying drawings, in which: 
         FIG. 1A  illustrates a post-filter, common gain determination type of automatic gain control system, according to an embodiment of the present invention; 
         FIG. 1B  illustrates another post-filter, common gain determination type of automatic gain control system, according to an embodiment of the present invention; 
         FIG. 1C  illustrates another post-filter, common gain determination type of automatic gain control system, according to an embodiment of the present invention; 
         FIG. 1D  illustrates an example plot of a transfer function that represents a Loudness Growth Function (LGF); 
         FIG. 2A  illustrates Gain Rules, e.g., of  FIG. 1A , in more detail, according to an embodiment of the present invention; 
         FIG. 2B  illustrates the Gain Rule, e.g., of  FIGS. 1B-1C and 4A  in more detail, according to an embodiment of the present invention; 
         FIGS. 2C-2D  illustrate continuous piece-wise linear, input-output functions that represent different examples of the operation of the Static Compression blocks of  FIGS. 12A-12B ; 
         FIG. 3A  illustrates a 0.6 second segment of the temporal waveform at the output of a Loudness Growth Function according to the Related Art; 
         FIG. 3B  illustrates a 0.6 second segment of the temporal waveform at the output of a Loudness Growth Function according to the an embodiment of the present invention; 
         FIG. 3C  illustrates a spectral profile at the output of a Loudness Growth Function according to the Related Art; 
         FIG. 3D  illustrates a spectral profile at the output of a Loudness Growth Function according to the an embodiment of the present invention; 
         FIG. 4A  illustrates another post-filter, common gain determination type of automatic gain control system, according to an embodiment of the present invention; 
         FIG. 4B  illustrates the Slow Gain rules of  FIG. 4A , in more detail; 
         FIG. 5  shows post-filter, common gain determination type of automatic gain control systems for a bilateral cochlear implant system, according to another embodiment of the present invention; 
         FIG. 6  shows post-filter, common gain determination type of automatic gain control systems for another bilateral cochlear implant system, according to another embodiment of the present invention; and 
         FIG. 7  illustrates a cochlear implant system, according to another embodiment of the present invention. 
     
    
    
     DETAILED DESCRIPTION 
     Embodiments of the present application are directed towards automatic gain control systems that feature post-filter (i.e., subsequent to bandpass filtering) gain generation, post-filter application of a common gain, and a static compression block configured with a maximum output level equal to a saturation level of a Loudness Growth Function. 
     A post-filter, common gain determination type of automatic gain control (AGC) system  1111 A, according to an embodiment of the present invention, is shown in  FIG. 1A . For simplicity of illustration, only four bands are shown in  FIG. 1A , but a higher number of bands is contemplated, e.g., 22 bands. Audio signal  1  is split into four frequency bands by four band-pass filters (BPFs)  11 - 14  (or, in other words, collectively, a frequency analysis unit). Each BPF passes a different band of frequencies. BPF outputs  21 - 24  are applied to amplitude detectors  31 - 34 , e.g., envelope detectors, to produce provisional amplitudes, e.g., provisional envelopes,  41 - 44 . Other examples of amplitude detectors include: full-wave rectifiers; half-wave rectifiers; peak detectors; quadrature envelope detection; etc. Provisional envelopes  41 - 44  are applied to gain rules  121 - 124  to generate provisional gains  131 - 134 . Provisional envelopes  41 - 44  also are applied to amplifiers  141 - 144  where they are adjusted so as to generate adjusted amplitudes, e.g., adjusted envelopes,  65 - 68 . Adjusted envelopes  65 - 68  are applied to loudness growth function (LGF) blocks  71 - 74  (or, in other words, a plurality of translation units) to produce magnitude signals  81 - 84 . 
     The components of system  1111 A can be discrete components or can be functional blocks implemented by, for example, a programmable processor, e.g., a digital signal processor (DSP). In the latter circumstance, e.g., filters  11 - 14  can be implemented by the processor performing a Fast Fourier Transformation (FFT) upon audio signal  1 . Another embodiment uses a quadrature pair of BPFs in each band, followed by quadrature envelope detection to produce the envelopes. 
     The bands have their own gain rules  121 - 124 , and produce their own provisional gains  131 - 134 , respectively. Rather than applying each of provisional gains  131 - 134  to its corresponding one of amplifiers  141 - 144 , respectively, as in the Related Art, i.e., using a band-specific gain technique, system  1111 A applies one common gain  201  to amplifiers  141 - 144 . This common gain  201  can be calculated by a Gain Combine block  211  based upon provisional gains  131 - 134 . An advantage of the common gain technique over the band-specific gain technique is that the spectral profile is better preserved. 
     More particularly, according to the band-specific gain technique, the AGC system on each frequency band operates independently of the AGC systems for the other bands. The band-specific gain technique is commonly used in hearing aids. For a hearing-aid wearer, hearing loss often varies with frequency, and thus it can be beneficial to apply differing amounts of compression in different frequency bands. However, for an AGC system that uses multiple bands, such a benefit is outweighed by the following drawback: because less gain is applied to intense bands than is applied to weak bands, the band-specific gain technique tends to reduce the amplitude of spectral peaks relative to spectral valleys, i.e., it flattens the spectral profile, which can degrade speech intelligibility. As an example, for a compression ratio of 4 or greater, speech intelligibility degrades as the number of channels is increased. See, e.g., the article by Plomp R (1994) “Noise, amplification, and compression: considerations of three main issues in hearing aid design,” Ear &amp; Hearing 15: 2-12. Plomp recommended using 2 to 4 channels, with a compression ratio of 2. Applying an AGC system using the band-specific gain technique AGC system with infinite compression to a cochlear implant system with, e.g., 22 channels would be expected to give very poor speech intelligibility. By contrast, the common gain technique applies the same gain to intense bands as is applied to weak bands, which avoids flattening the spectral profile, i.e., which better preserves the spectral profile, and so achieves relatively better speech intelligibility. 
     Operation of system  1111 A can be described as filters  11 - 14  performing a frequency analysis to decompose audio signal  1  into analysis signals  21 - 24  contained in frequency bands, respectively. Envelope detectors  31 - 34  produce provisional envelopes  41 - 44  based upon analysis signals  21 - 24 , respectively. Provisional gains  131 - 134  are generated by gain rules  121 - 124  based upon provisional envelopes  41 - 44 , respectively. A common gain is determined by gain combine block  211  based upon provisional gains  131 - 134 . And the updated common gain  201  is applied to provisional envelopes  41 - 44  by amplifiers  141 - 144  to produce adjusted envelopes  65 - 68 . 
     In one embodiment, Gain Combine block  211  calculates the minimum of the provisional gains  131 - 134 . In another embodiment, the Gain Combine block  211  calculates the median of the provisional gains  131 - 134 . In another embodiment, the Gain Combine block  211  calculates the weighted mean of the provisional gains  131 - 134 . All bands may be given equal weight, or alternatively different weights may be applied to different bands. For example, more weight may be given to bands that are more important for speech intelligibility.  FIG. 1A  can be summarized as illustrating post-filter gain generation and a post-filter application of a common gain. 
     Another post-filter, common gain determination type of automatic gain control (AGC) system  1111 B, according to an embodiment of the present invention, is shown in  FIG. 1B . In contrast to system  1111 A, system  1111 B has a single gain rule  241 , which is common to all bands. Provisional amplitudes, e.g., provisional envelopes,  41 - 44  are applied to Level Combine block  221 , which determines a single level  231 . Level  231  is applied to Gain Rule  241 , to produce common gain  201 . In one embodiment, Level Combine block  221  calculates the maximum of the individual envelopes  41 - 44 . In yet another embodiment, Level Combine block  221  calculates the median of the individual provisional envelopes  41 - 44 . In another embodiment, Level Combine block  221  calculates the weighted mean of the individual provisional envelopes  41 - 44 . All bands may be given equal weight, or alternatively different weights may be applied to different bands. For example, more weight may be given to bands that are more important for speech intelligibility.  FIG. 1B  can be summarized as illustrating post-filter gain generation and a post-filter application of a common gain. 
       FIG. 1C  illustrates another post-filter, common gain determination type of AGC system  1111 C according to an embodiment of the present invention. System  1111 C incorporates temporal fine structure (e.g., which may improve prove pitch perception) in addition to post-filter gain generation and application of a common gain as in system  1111 B of  FIG. 1B . Likewise  FIG. 1C  can be summarized as illustrating post-filter gain generation and a post-filter application of a common gain. 
     In  FIG. 1C , system  1111 C is shown as a four-band system for simplicity of illustration, but a higher number of bands (for example 22) is more typical. In comparison to system  1111 B, the BPF outputs  21 - 24  are processed on two paths: an amplitude path and a timing path. The amplitude path comprises Amplitude Detectors  31 - 34  and LGF blocks  71 - 74  and is similar to the processing in system  1111 B. The timing path comprises Timing Detectors  401 - 403 , which generate timing signals  411 - 414 . Pulse Generator  281  uses both magnitude signals  81 - 84  and timing signals  411 - 414  to generate the stimulation pulse data  291 . Generally, the magnitude signals  81 - 84  determine the current levels of the stimulation pulses. 
     The LGF blocks  71 - 74  of  FIGS. 1A-1C  perform instantaneous non-linear compression. Generally a logarithmic or power-law transfer function is used.  FIG. 1D  illustrates an example of a non-linear compression transfer function that can be used to implement the LGF blocks  71 - 74 . In  FIG. 1D , amplitudes equal to a specified saturation level are mapped to magnitude value of 1.0, which will result in C-level stimulation. The saturation level is often taken as a reference point, e.g., labeled as 0 dB. Envelope amplitudes greater than the saturation level are clipped to magnitude value 1.0. Envelope amplitudes equal to a specified base level are mapped to magnitude value 0.0, which will result in T-level stimulation. The dynamic range is defined as the ratio of the saturation level to the base level. Typical dynamic range values are from 30 to 50 dB;  FIG. 1D  shows a dynamic range, e.g., of 40 dB. 
     The LGF blocks  71 - 74  reduce (if not prevent) excessive loudness by limiting the current on a channel to C-level. However, if the amplitudes provided to the LGF are permitted to exceed the saturation level, then clipping occurs. Clipping has undesirable effects that include the following. Firstly, it can distort the temporal waveform of the envelopes, reducing modulation depth. Secondly, as the channel with largest amplitude will clip first, it can reduce the ratio of the spectral peaks to the spectral valleys, flattening the spectral profile and distorting formant patterns. Thirdly, in the presence of background noise, the speech signal will tend to clip more often than the noise, reducing the effective signal-to-noise ratio (SNR). Clipping can be reduced (if not minimized) by, e.g., appropriate configuration of the Gain Rule, as discussed below. 
     Gain Rules  121 - 124  in  FIG. 1A  and common Gain Rule  241  in  FIGS. 1B-1C  can be configured, e.g., with similar (or the same) internal architectures.  FIG. 2A  shows each of gain rules  121 - 124  in more detail, according to another embodiment of the present invention. For example, in terms of gain rule  121 ,  FIG. 2A  illustrates provisional amplitude  41  as the input signal (which is provided to Level Dynamics bock  1201 ) and provisional gain  131  as the output signal.  FIG. 2B  shows common Gain rule  241  in more detail, according to another embodiment of the present invention. The input signal ( 41 ,  42 ,  43 ,  44  or  231 , respectively) is applied to a Level Dynamics block  1201  to generate a processed level  1202 . A Static Compression block  1203  uses processed level  1202  to determine a raw gain  1204 , which is further processed by a Gain Dynamics block  1205  to produce the output gain ( 131 ,  123 ,  133 ,  134  or  201 , respectively). 
     The operation of the Static Compression block  1203  can be described by an input-output function. The input-output function can be, e.g., a continuous piece-wise linear function, specified by two or more compression ratios and a corresponding number of knee points. Examples of continuous, piece-wise linear input-output functions that can be used to implement Static Compression block  1203  are illustrated in  FIGS. 2C-2D . The compression ratio can be defined, e.g., as the change in input level that produces a 1 dB change in output level, i.e., the reciprocal of slope of the input-output function. In  FIG. 2C , for input levels up to a knee-point of 70 dB, the output level is the same as the input level. This region has a compression ratio of 1, i.e., linear amplification. For input levels above 70 dB, the output level remains at 70 dB, which is the maximum output level of the embodiment reflected in  FIG. 2C . This region has infinite (or substantially infinite) compression, hence the corresponding knee-point in this embodiment may be referred to as an infinite compression knee-point.  FIG. 12D  is an example of an input-output function with two knee-points. For input levels up to a first knee-point of 30 dB, the output level is the same as the input level. For input levels in the range 30 dB up to a second knee-point of 70 dB, the output level grows half as much as the input level. This region has a compression ratio of 2 (i.e., 2:1 compression). For input levels above 70 dB, the output level remains at 50 dB, which is the maximum output level of the embodiment reflected in  FIG. 2D . Unlike conventional gain rules, static compression block  1203  in each of  FIGS. 2A-2B  is configured with a maximum output level equal to the LGF saturation level. This reduces, if not eliminates, clipping in LGF blocks  71 - 74 . In some embodiments, e.g., the embodiment reflected in  FIG. 2C , the infinite compression knee point is equal to the maximum output level. In other embodiments, e.g., the embodiment reflected in  FIG. 2D , it is not. 
     An embodiment of Level Combine block  221  and Gain Rule  241  can be summarized as:
         Level Combine: maximum level.   Static Compression: linear amplification up to a knee-point equal to the LGF saturation level, then infinite compression for higher levels.   Level Dynamics: none, i.e. zero attack time.   Gain Dynamics: a hold time of 200 ms, followed by a release period where the gain increases at a constant slew-rate of 40 dB per second.       

     An example of MATLAB code that can be used to implement Level Combine block  221  and Gain Rule  241  is: 
     
       
         
               
               
               
             
           
               
                   
                   
               
             
             
               
                   
                   
                    %% Initialization: 
               
               
                   
                   
                    % Configuration parameters: 
               
               
                   
                   
                    sample_ rate= 1 000; %Hz 
               
               
                   
                   
                    saturation _level= 1.0; 
               
               
                   
                   
                    slew_ rate = 40; % dB/sec 
               
               
                   
                   
                    hold time = 0.2;% seconds 
               
               
                   
                   
                    max_gain = 1.0; 
               
               
                   
                   
                    %% Parameter calculations: 
               
               
                   
                   
                    step_ dB = slew _rate I sample rate; 
               
               
                   
                   
                    scaler = 10Λ(step_dB/20); 
               
               
                   
                   
                    hold_ count= hold_ time * sample rate; 
               
               
                   
                   
                    % State variables: 
               
               
                   
                   
                    gain= max _gain; 
               
               
                   
                   
                    held=O; 
               
               
                   
                   
                    %Processing: 
               
               
                   
                   
                    %Level Combine: 
               
               
                   
                   
                    largest_ env =max( envelopes); 
               
               
                   
                   
                    %Gain rule: 
               
               
                   
                   
                    raw _gain= saturation _level I largest_ env; 
               
               
                   
                   
                    if raw _gain&lt; gain 
               
               
                   
                   
                    %Attack 
               
               
                   
                   
                    gain= raw _gain; 
               
               
                   
                   
                    held= 0; % Start hold timer 
               
               
                   
                   
                   else 
               
               
                   
                   
                    held= held+ 1; 
               
               
                   
                   
                    if held&gt; hold count 
               
               
                   
                   
                    %Release 
               
               
                   
                   
                    gain = gain * scaler; 
               
               
                   
                   
                   else 
               
               
                   
                   
                    %Hold 
               
               
                   
                   
                  end 
               
               
                   
                   
                 end 
               
               
                   
                   
                 gain= min( max _gain,gain); 
               
               
                   
                   
               
             
          
         
       
     
     A benefit of at least some embodiments of the present invention is shown by the contrast between  FIG. 3A  (representative of Related Art) vis-a-vis  FIG. 3B  (representative of an embodiment of the present invention), and by the contrast between  FIG. 3C  (representative of Related Art) vis-a-vis  FIG. 3D  (representative of an embodiment of the present invention). Here, for example, an audio signal in the form of a sentence in the presence of background noise is considered albeit for 22 bands, not merely 4 bands.  FIGS. 3A  (Related Art) and  3 B (present embodiment) show a 0.6 second segment of the temporal waveform at the output of the LGF, e.g., for channel  4  (centered at 625 Hz), of a 22-channel system. Related Art  FIG. 3A  shows the LGF output signal  1302  for a Related Art system  100  utilizing a pre-filter gain determination type of AGC. In Related Art  FIG. 3A , as called out by reference  1304 , the signal  1302  is clipped over the time interval of approximately 0.33 to 0.39 seconds.  FIG. 3B , by contrast, shows the corresponding output signal  1306  according to an embodiment of the present invention, e.g., systems  1111 A and  1111 B. As indicated by reference  1308 , no clipping occurs. Relative to Related Art  FIG. 3A ,  FIG. 3B  (present embodiment) shows that more of the amplitude modulation, which is a cue to the voice pitch, is preserved. 
       FIGS. 3C  (representative of Related Art) and  3 D (representative of an embodiment of the present invention) extend the examples of  FIGS. 3A-3B  by showing spectral profiles  1310  and  1312  at the output of the LGF blocks, respectively, albeit for the 22 channels, e.g., at the time 0.36 seconds approximately. In  FIG. 3D , the spectral profile  1312  shows at most that one channel (in this case channel  6 ) reaches magnitude 1.0 and produces stimulation at C-level on the corresponding electrode. This gives a clearer indication of the first formant frequency (the first peak at channel  6  in the spectral profile  1312 ). In contrast, the spectral profile  1310  of  FIG. 3C  (which, again, is produced by a pre-filter gain determination type of AGC system according to the Related Art) shows that clipping occurs for channels  4 ,  5 ,  6 , and  7 , i.e., those four channels have the maximum magnitude (1.0) resulting in stimulation at C-level on the corresponding electrodes. But for the clipping, a peak would be apparent on one of channels  4 - 7 . Due to the clipping, however, it is unclear which one of the channels  4 - 7  has the peak; consequently, the frequency of the first formant cannot be accurately determined from the spectral profile  1310 . Relative to Related Art  FIG. 3C ,  FIG. 3D  (present embodiment) shows improved speech intelligibility. 
     Another embodiment, according to the present invention, of a post-filter, common gain determination type of AGC system  1114  is shown in  FIG. 4A . In this arrangement, provisional amplitudes, e.g., envelopes,  41 - 44  are processed by Slow Gain Modules  301 - 304  to produce processed amplitudes, e.g., envelopes,  311 - 314 . A Level Combine block  221 A receives processed envelopes  311 - 314 , determines a maximum one thereof, and outputs the maximum as level  231 A to a Fast Gain rule  241 A, which then produces common gain  201 . Fast Gain Rule  241  is implemented, e.g., as in the MATLAB code listed above. Slow Gain Modules  301 - 304  act independently. A purpose of Slow Gain Modules  301 - 304  is to help transition from one environment to the next, e.g., to compensate for differences in environment, such as between one talker and another talker, or between a quiet room and a noisy street. This is sometimes known as an automatic volume control (AVC).  FIG. 4A  can be summarized as illustrating post-filter gain generation and a post-filter application of a common gain. 
     Slow Gain Modules  301 - 304  can be configured with similar (or the same) internal architectures.  FIG. 4B  shows each of Slow Gain Modules  301 - 304  in more detail. The input signal ( 41 ,  42 ,  43  or  44 , respectively) is applied to a variable-gain amplifier  1404  to produce the processed amplitudes ( 311 ,  312 ,  313  or  314 , respectively). This operation is equivalent to multiplying the input signal by a gain  408 . A level detector  1405  produces a signal  1406 , which represents the level of the input signal. Generally, the level detector  1405  rectifies and smoothes the input signal. A gain rule  1407  uses signal  1406  to determine the gain  1408 . Alternatively, the Slow Gain Modules can be implemented using, e.g., the Adaptive Dynamic Range Optimization (ADRO) technique, as disclosed in U.S. Pat. No. 6,731,767 B1 by Blarney et al. 
     The time taken for an AGC system to respond to an increase in input level is called the attack time. The time taken for an AGC system to respond to a subsequent decrease in input level is called the release time. Typical settings for a “fast” AGC are an attack time in the range of 2 to 5 ms, and a release time in the range 7 5 to 300 ms. The attack and release times should be selected so that the gain changes are small over the course of a sentence. Suitable attack times are in the range 0.5-1 second, and suitable release times are in the range 1-2 seconds. 
       FIG. 5  shows post-filter, common gain determination type of automatic gain control systems  1000  and  2000  for a bilateral cochlear implant system  1115 , according to another embodiment of the present invention. The bilateral system contains two systems,  1000  and  2000 , which will be referred to as Left system  1000  and Right system  2000 . Each system is similar to system  1111 B, except that Left gain  1006  and Right gain  2006  are provided, e.g., to a minimum (Min) block  1007  in Left system  1000 , which calculates common gain  1008  as the minimum of Left gain and Right gain  2006  at each instant in time. Alternatively, Min block  2007  can be provided in Right system  2000 . Common gain  1008  is applied to both Left amplifiers  1021 - 1024  and Right amplifiers  2021 - 2024 . Each Gain Rule ( 1005 ,  2005 ) has a maximum output level equal to the LGF saturation level.  FIG. 5  can be summarized as illustrating post-filter gain generation and a post-filter application of a common gain. 
       FIG. 6  shows post-filter, common gain determination type of automatic gain control systems  1000 ′ and  2000 ′ for another bilateral cochlear implant system  1116 , according to an embodiment of the present invention. Systems  1000 ′ and  2000 ′ are similar to systems  1000  and  2000  of  FIG. 5 , respectively, except that Left maximum envelope  1004  and Right maximum envelope  2004  are provided to a maximum (Max) block  1009 , which calculates the overall maximum envelope  1010 . This is used by Gain Rule  1005  to generate common gain  1008 , which is applied to both left amplifiers  1021 - 1024  and Right amplifiers  2021 - 2024 . In  FIG. 6 , Max block  1009  and Gain Rule  1005  are illustrated as being included within Left system  1000 ′; alternatively, provided in Max block  1009  and Gain Rule  1005  can be provided in Right system  2000 ′.  FIG. 6  can be summarized as illustrating post-filter gain generation and a post-filter application of a common gain. 
     A benefit of bilateral hearing is the ability to localize sound. One cue that is used in localization is the interaurallevel difference (ILD). For example, a sound coming from the left side will have a greater intensity at the left ear than the right ear. Disadvantages of clipping in AGC systems, e.g.,  1111 B, have been discussed above. In the context of bilateral cochlear implant systems, clipping has further disadvantages. If clipping occurs on one or both sides, then the ILD cue is reduced or destroyed. However, systems  1115  and  1116 , like system  1111 B, avoid clipping, thereby better preserving the ILD cue and facilitating better sound localization by the recipient. 
     Some embodiments of the present invention may be implemented in sound processing technologies, for example, hearing prostheses, e.g., cochlear implant systems.  FIG. 7  illustrates a perspective view of a cochlear implant system  1117  according to another embodiment of the present invention. System  1117  includes a sound processor module  126  which can include any of gain control systems  1111 A,  1111 B,  1111 C or  1114 , or if system  117  is part of a bilateral cochlear implant system, then corresponding portions of gain control systems  1115  or  1116 , according to embodiments of the present invention, respectively. 
     In  FIG. 7 , cochlear implant system  1117  is illustrated as implanted in a recipient having an outer ear  101 , a middle ear  105  and an inner ear  107 . Components of outer ear  101 , middle ear  105  and inner ear  107  are described below, followed by a description of cochlear implant  100 . 
     In a fully functional ear, outer ear  101  comprises an auricle  110  and an ear canal  102 . An acoustic pressure or sound wave  103  is collected by auricle  110  and channeled into and through ear canal  102 . Disposed across the distal end of ear cannel  102  is a tympanic membrane  104  which vibrates in response to sound wave  103 . This vibration is coupled to oval window or fenestra ovalis  112  through three bones of middle ear  105 , collectively referred to as the ossicles  106  and comprising the malleus  108 , the incus  109  and the stapes  111 . Bones  108 ,  109  and  111  of middle ear  105  serve to filter and amplify sound wave  103 , causing oval window  112  to articulate, or vibrate in response to vibration of tympanic membrane  104 . This vibration sets up waves of fluid motion of the perilymph within cochlea  140 . Such fluid motion, in tum, activates tiny hair cells (not shown) inside of cochlea  140 . Activation of the hair cells causes appropriate nerve impulses to be generated and transferred through the spiral ganglion cells (not shown) and auditory nerve  114  to the brain (also not shown) where they are perceived as sound. 
     Cochlear implant  100  comprises an external component  142  which is directly or indirectly attached to the body of the recipient, and an internal or implantable component  144  which is temporarily or permanently implanted in the recipient. External component  142  typically comprises one or more sound input elements, such as microphone  124  for detecting sound, a sound processing unit  126 , a power source (not shown), and an external transmitter unit  128 . External transmitter unit  128  comprises an external coil  130  and, preferably, a magnet (not shown) secured directly or indirectly to external coil  130 . Sound processing unit  126  processes the output of microphone  124  that is positioned, in the depicted embodiment, by auricle  110  of the recipient. Sound processing unit  126  generates encoded signals, sometimes referred to herein as encoded data signals, which are provided to external transmitter unit  128  via a cable (not shown). As shown by exploded view  186  in  FIG. 7 , sound processor module  126  can include a programmable processor  190 , e.g., a digital signal processor (DSP), application-specific integrated circuit (ASIC), etc. Processor  190  is operatively coupled to a memory  192 , e.g., random access memory (RAM) and/or read-only memory (ROM). Processor  192  also is operatively coupled via interface  188 , e.g., to a microphone  124  and external transmitter unit  128 . 
     Internal component  144  comprises an internal receiver unit  132 , a stimulator unit  120 , and an elongate stimulating lead assembly  118 . Internal receiver unit  132  comprises an internal coil  136 , and preferably, a magnet (also not shown) fixed relative to the internal coil. Internal receiver unit  132  and stimulator unit  120  are hermetically sealed within a biocompatible housing, sometimes collectively referred to as a stimulator/receiver unit. Internal coil  136  receives power and stimulation data from external coil  130 , as noted above. Elongate stimulating lead assembly  118  has a proximal end connected to stimulator unit  120 , and extends through mastoid bone  119 . Lead assembly  118  has a distal region, referred to as electrode assembly  145 , implanted in cochlea  140 . As used herein the term “stimulating lead assembly,” refers to any device capable of providing stimulation to a recipient, such as, for example, electrical or optical stimulation. 
     Electrode assembly  145  may be implanted at least in basal region  116  of cochlea  140 , and sometimes further. For example, electrode assembly  145  may extend towards apical end of cochlea  140 , referred to as cochlea apex  134 . Electrode assembly  145  may be inserted into cochlea  140  via a cochleostomy  122 , or through round window  121 , oval window  112 , and the promontory  123  or opening in an apical tum  147  of cochlea  140 . 
     Electrode assembly  145  has disposed therein or thereon a longitudinally aligned and distally extending array  146  of electrode contacts  148 , sometimes referred to as electrode array  146  herein. Throughout this description, the term “electrode array” means a collection of two or more electrode contacts, sometimes referred to simply as contacts herein. As would be appreciated, electrode array  146  may be disposed on electrode assembly  145 . However, in most practical applications, electrode array  146  is integrated into electrode assembly  145 . As used herein, electrode contacts or other elements disposed in a carrier refer to elements integrated in, or positioned on, the carrier member. As such, electrode array  146  is referred to herein as being disposed in electrode assembly  145 . Stimulator unit  120  generates stimulation signals which are applied by electrodes  148  to cochlea  140 , thereby stimulating auditory nerve  114 . 
     In cochlear implant  100 , external coil  130  transmits electrical signals (i.e., power and stimulation data) to internal coil  136  via a radio frequency (RF) link. Internal coil  136  is typically a wire antenna coil comprised of multiple turns of electrically insulated single-strand or multi-strand platinum or gold wire. The electrical insulation of internal coil  136  is provided by a flexible silicone molding (not shown). In use, implantable receiver unit  132  may be positioned in a recess of the temporal bone adjacent auricle  110  of the recipient. 
     As noted,  FIG. 7  illustrates specific embodiments of the present invention in which cochlear implant  100  includes an external component  142 . It would be appreciated that in alternative embodiments, cochlear implant  100  comprises a totally implantable prosthesis that is capable of operating, at least for a period of time, without the need of an external component. In such embodiments, all components of cochlear implant  100  are implantable, and the cochlear implant operates in conjunction with external component  142 . 
     Some embodiments of the present invention are described herein in connection with a type of Active Implantable Medical Device (AIMD), namely a cochlear implant system. It should be appreciated that embodiments of the present invention may be implemented in other sound-processing technologies that benefit from gain control systems, e.g., telecommunications, and the like. 
     Throughout the specification and the claims that follow, unless the context requires otherwise, the words “comprise” and “include” and variations such as “comprising” and “including” will be understood to imply the inclusion of a stated integer or group of integers, but not the exclusion of any other integer or group of integers. 
     Reference herein to “one embodiment” or “an embodiment” means that a particular feature, structure, operation, or other characteristic described in connection with the embodiment may be included in at least one implementation of the present invention. However, the appearance of the phrase “in one embodiment” or “in an embodiment” in various places in the specification does not necessarily refer to the same embodiment. It is further envisioned that a skilled person could use any or all of the above embodiments in any compatible combination or permutation. 
     While various embodiments of the present invention have been described above, it should be understood that they have been presented by way of example only, and not limitation. It will be apparent to persons skilled in the relevant art that various changes in form and detail may be made therein without departing from the scope of the present invention. Thus, the breadth and scope of the present invention should not be limited by any of the above-described exemplary embodiments, but should be defined only in accordance with the following claims and their equivalents.

Technology Classification (CPC): 7