DE29910036U1 - phone - Google Patents

Description

phone

The invention relates to a telephone with a memory for storing acoustic signals for the purpose of reproducing them as a call signal tone.

A telephone of this type is disclosed in the German patent application 198 46 544 of the same applicant. The possibility of storing and reproducing individual acoustic signals as a call signal tone has been particularly

The advantage of mobile phones is that everyone can immediately tell from the ring tone whether a call is for them or from another mobile phone. While with normal ring signals a tone generator generates vibrations in the audible frequency spectrum, which can then be converted into an acoustic signal by a loudspeaker, this is not possible when playing back individual acoustic signals. Instead, these signals must be recorded or stored in as natural a form as possible and read out or played back to signal a call. The problem here is how to achieve as natural a reproduction of the recorded acoustic signal as possible.

This problem can be solved in a telephone with a memory for storing acoustic signals for the purpose of reproducing them as a call signal tone using a module for processing an acoustic signal before it is written into the memory and/or after it is read out from it. With these measures, the recorded acoustic signal can be adapted to the special features of the telephone's internal storage and playback system in order to achieve the most natural sound reproduction possible despite the limited properties of mobile phones with regard to sound transmission.

Within the scope of the invention, a wide variety of modifications of the acoustic signal are possible, with the most appropriate modifications depending on the

structural conditions of the telephone set and/or the type of acoustic signal to be reproduced. Although the acoustic signal could also be read out of the memory for processing and re-written after the relevant modification, this method is comparatively complex and is therefore not preferred by the invention. Rather, the acoustic signal should be modified directly during recording and/or only during playback. A distinction must be made here between internal processing, which does not distort the acoustic signal but only serves to simplify the internal signal processing, and

&iacgr;&ogr; actual change in the acoustic signal in the manner of a pre-distortion, so that during playback a sound signal is generated that is as close as possible to the original signal due to the design-related distortion.

Two trends have emerged in modern electronics, namely analog signal processing and digital signal processing. While in the former the electrical signals are represented as voltage signals with continuously variable amplitude, in digital signal processing the original signal is not only sampled over time and thus converted into a step function, but the amplitude value is also mapped to a range of possible digital numbers, whereby quantization takes place with regard to the amplitude. If the quantization steps are chosen to be sufficiently small, the original signal can be reproduced from such a digital signal with sufficient precision, while the storage and especially processing of a digital signal offers many advantages over processing the analog signal. If the acoustic signal is stored in the form of digital values, it is advisable to carry out pre-processing and, if necessary, post-processing on this digital level. In this case, the processing module must be coupled with the acoustic signal converted into a digital signal.

As part of such a procedure, it has proven useful to provide means for compressing and/or decompressing the acoustic signal in the processing module. As part of the storage of electronic signals, it is known to sample the original, continuous-time signal at regular intervals and then to keep the relevant amplitude values constant, to record them and to store them either directly as a voltage value or indirectly as a digital value. In order to be able to reproduce the original signal from the values stored in this way with as little distortion as possible, the

The sampling theorem must be observed, according to which the sampling frequency must be twice as high as the highest frequency of the acoustic frequency spectrum that can still be reproduced. This goes up to about 20 kHz, so that the theoretically still reasonable maximum sampling frequency is about 40 kHz. This would mean that 40,000 voltage or digital values would have to be stored per second of the acoustic signal to be recorded. This enormous amount of data can only be reduced by two measures: firstly, by reducing the sampling rate, which means that high frequencies can no longer be recorded, which, however, is not very noticeable in view of the mediocre reproduction quality of telephones, provided that the sampling rate does not fall below a frequency of about 20,000 to 25,000 Hz. A further reduction in the amount of data can then only be achieved by data compression. Various measures have become known for this purpose. For example, storage can be interrupted during pauses in the acoustic signal by storing only the length of the pause as information. In the case of approximately constant signals, which are characterized by a sequence of multiple, periodic signals, the continuous recording of the acoustic signal can also be interrupted and replaced, for example, by storing the start and end addresses of the signal to be repeated and the number of repetitions. Other compression techniques include, for example, the transformation of an acoustic signal of limited length into a frequency spectrum, the subsequent filtering out

of those frequency components with a significant amplitude component, which can be accomplished, for example, by comparing them with a threshold value, and then storing the frequency components filtered out in this way with the relevant amplitude value. Other compression techniques are conceivable, but will not be discussed in detail here. A compression of the data signal to be stored corresponds to a

Decompression function of the read data in order to at least approximately recreate the original signal. This decompression can possibly be carried out in a second processing module,

&iacgr;&ogr; may also be carried out by the same processing module provided that it is able to perform the different functions.

As already explained above, data compression can be achieved in different ways, and the design of the processing module(s) required for this depends on the desired function. The filtering out of "silent" intervals in the acoustic signal can be achieved using comparators, which compare the amplitude of the acoustic signal with constant positive and/or negative threshold values. To filter out phases with an almost unchanged sound signal, the last stored signal period can be output with an unchanged time scale and subtracted from the current value of the acoustic signal, and the difference signal is again checked using comparators to see whether it is approximately zero, for which constant threshold values can be used. Finally, digital or fast Fourier transformation can be used to transform the time signal into the frequency range. A corresponding transformer can be implemented using signal processors.

The aim of all these measures is to reduce the amount of memory required to store a signal as much as possible in order to be able to record longer acoustic signals using a memory of standard size.

or to be able to increase the sampling rate so that the playback quality is improved, especially in the upper range of the frequency spectrum.

Within the scope of the invention, however, it is also possible for the processing module to be coupled to the acoustic signal in its analog form. In such a case, the processing module has the form of a filter or other circuit constructed with analog components, the output signal of which is in turn an analog signal. This embodiment allows in particular the implementation of the inventive principle without digitizing the acoustic signal and can therefore be used to reduce the design effort when memory modules are used which are able to record analog voltage values. Such modules can be implemented in particular by means of capacitive charge storage.

Both in the context of processing digitized signals and in analog signal processing, it is possible to construct the processing module in the form of a filter, with means for selectively changing the amplitude of various frequency ranges of the acoustic signal. In this case, it is particularly possible to emphasize frequency components of the acoustic signal that are excessively attenuated by the recording device and/or emitted by the playback device only with reduced amplification by means of greater amplification, so that these signal components are also reproduced approximately to their original extent. Other frequencies at which the recording and/or playback device exhibits resonant behavior can be attenuated electronically in order to prevent an unrealistic overestimation of these frequency components. The aim of these measures is to achieve a linearization of the transmission frequency response of the overall system for the largest possible frequency spectrum of the original signal.

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Further advantages arise from the fact that the processing module contains means for frequency shifting the acoustic signal. For some signals, it is possible to raise the entire frequency spectrum to a higher frequency range without unduly distorting the character of the acoustic signal. In such cases, it is sensible to shift the frequency components characteristic of an acoustic signal to the most favorable frequency range for the overall system and thus to retain the character of the acoustic signal as far as possible despite a reduction in the frequency spectrum. In order to achieve the best possible result,

To obtain the best possible sound quality, it is recommended that as large a frequency spectrum as possible of an original signal is shifted to the frequency range where the entire playback system has maximum amplification and/or the human ear is most sensitive. The extent to which the original acoustic signal is to be shifted along the frequency axis can be individually determined by playing the "shifted" signal and setting the difference or shift frequency higher or lower based on the sound result.

Within the scope of the invention, an analog memory can be advantageously used, into which the possibly processed acoustic signal is written. As already explained above, the design effort can be reduced by this measure.

Further advantages are provided by an input option for selecting one or more different modifications of the original signal, for example by selecting from a menu shown on a display element of the telephone. It is conceivable to browse through the various options in a "modification menu" and select them by pressing a "Yes" key or hide them with a "No" key. Such menu items can be used, for example,

be: "Bass boost", "Mid boost", "Treble boost", "Frequency boost", "Data compression". Additional parameters can be specified within the individual menu items by selecting the

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The relevant parameters can be increased using a "greater" button or decreased using a "lesser" button, e.g. the degree of amplification of bass, mids and/or treble or the degree of frequency shift. If necessary, you can also choose between different types of data compression.

The invention further provides an amplifier which acts directly on the signal read out from the storage element and possibly modified, e.g. decompressed, and which is connected on the output side to an electroacoustic

γ converter. According to this inventive concept, a reconversion of the digital signal can be dispensed with if instead another coding is carried out, for example as part of a pulse duration modulation corresponding to the current digital value, so that the corresponding information is not contained in the amplitude of a (reconverted) voltage value, but in the pulse widths of a square wave signal alternating between an upper and a lower voltage value. This measure can bring advantages in particular when implementing the necessary amplifier, since in pure switching operation there are hardly any power losses and thus the limited available energy of an accumulator can be converted almost completely into sound waves.

The latter principle can be used with particular advantage if a smoothing element for the amplified signal is present in or in front of the electroacoustic converter. Such a smoothing element can be implemented in particular by means of a large inductance, as is already present in loudspeaker coils. Within the framework of such an inductance, the current flow will only change slowly even in a clocked switching operation of the voltage, so that the harmonics caused by the switching operation are not converted into an audible frequency.

Further features, details, advantages and effects based on the invention will become apparent from the following description of a

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preferred embodiment of the invention and the drawing. This shows in:

Figure 1 shows a first embodiment of the telephone according to the invention;

as well as

Figure 2 shows a modified embodiment, wherein the

Acoustic signal is digitized.

&iacgr;&ogr; Within the scope of the recording/playback system 1 according to Figure 1, the principle according to the invention can be implemented with minimal effort. A microphone 2, which is arranged, for example, in the mouthpiece of a preferably mobile telephone, is followed by a sample-and-hold element H. This transmits the respective microphone output signal 3 to the output 4 of the module H in a fixed time grid, while the output signal 4 is kept constant between these "sensitive" points in time. The time-discrete signal 4 is read into an analog memory M at the same time as these samples. Such analog memories work, for example, with charge quantities that are stored in capacitive elements and are available, for example, from the company ISD, San Jose, USA. Analog voltage signals can be stored in this way for a period of about a hundred years.

When a telephone call arrives, the acoustic signal stored in the memory M is read out 5 and transmitted to a processing module F that influences the frequency. This processing module F can either be implemented as a filter in order to transmit certain frequencies in an attenuated form or preferentially, or it can be designed as an analog multiplier in order to generate the product of the original signal 5 and an offset signal corresponding to the frequency shift in the manner of a modulator. Such a multiplication creates a frequency mixture with two sidebands that are symmetrically upper and lower.

below the offset or carrier frequency. By filtering out the lower sideband, for example, the original acoustic signal can be transferred to a higher frequency range. The signal 6 filtered or frequency-shifted in this way is made available by a downstream amplifier module 7 in a low-impedance manner in order to generate a sufficiently high sound pressure at the loudspeaker 8 that is clearly perceptible at a range of 10 m, for example.

The function of system 1 is controlled by a control module S, which

&iacgr;&ogr; can be implemented in particular as a microprocessor. This communicates on the one hand with the memory module M in order to initiate and abort the write or read process, and also with the modification module F in order to determine its function and/or to specify the required parameters.

The control module S receives its information via the telephone’s own keypad

T. It also communicates with an alphanumeric display device A, in order to offer the phone owner various parameters or commands for selection or specification, for example. In conjunction with the display unit A and the keyboard T, a large number of modification commands and/or parameters can be entered using just a few hardware elements within a menu-driven structure, which are registered by the control unit S and converted into corresponding control commands for the memory M and the modification module F.

The control of the system 11 according to Figure 2 is carried out according to the same principle as in the arrangement 1 according to Figure 1, namely via a control module S with which the telephone owner communicates via a display device A and a keyboard T, and which acts on a memory module M on the one hand and on a modification module F on the other.

The signal source, the microphone 2, and the electroacoustic transducer in the form of the loudspeaker 8 are also identical to the embodiment according to Figure 1.

In contrast to the circuit 1, however, in the system 11 the staircase signal 4 generated by the sample-and-hold element H from the microphone signal 3 is not written directly into an analog memory, but is converted by an AJD converter into a digital signal 12, which is written 13 into the digital memory M as a digital value after data compression in the processing module K.

In the case of a telephone call, the data stored in the digital memory M is compressed

The acoustic signal stored in the form is read out 14 and converted at least approximately into a digital image 15 of the original acoustic signal in a downstream decompression module D. This digital signal sequence 15 is transformed by a downstream DFT module by means of the digital Fourier transformation into a spectral sequence 16, which is acted upon by a digital filter F, for example in order to shift the elements of the spectral sequence 16 in terms of frequency or to emphasize or suppress individual frequency components. The spectral sequence 17 modified in this way is then converted back into a time sequence 18 within the framework of the inverse, digital Fourier transformation DFT" ¹ and converted in a downstream pulse duration modulation module PDM into a square-wave signal 19 alternating between two voltage values, the duration of the pulses each containing the information for the relevant digital value of the time sequence 18. This pulse duration modulated signal 19 is decoupled by a switching amplifier 20 in order to be able to drive the loudspeaker 8 connected on the output side with sufficient power. As a result of the square-wave sequence 19 alternating between two voltage levels, the output signal 21 of the amplifier 20 can be switched back and forth between a positive and a negative limit level, for example, so that hardly any power loss occurs in the amplifier 20 and almost all of the energy taken from the accumulator of the mobile telephone device can be fed to the loudspeaker 8. To ensure that the switching frequency of the pulse sequence 21 cannot be detected in the emitted acoustic signal,

A smoothing element G, e.g. in the form of an inductance, must be connected between the amplifier 20 and the loudspeaker 8. In some cases, the coil of the loudspeaker 8 can also take on this function. If necessary, the frequency of the square-wave signal 19 can also be shifted to the ultrasonic range.

The functions of the data compressor module K, the data decompressor D, the digital Fourier transformer DFT and the inverse DFT transformer DFT ¹ can be carried out by a fast signal processor 22, which further reduces the number of components required. The function of the modification module F can either be taken over by a digital module, e.g. a shift register or a digital filter, or can be transferred to the area of responsibility of the control module S or the signal processor 22.

Claims (13)

Protection claims 1. Telephone with a memory (M) for storing acoustic signals for the purpose of playing them back as a call signal tone, characterized by a module for processing an acoustic signal before writing it into the memory (M) and/or after reading it out from the same. 2. Telephone according to claim 1, characterized in that the &iacgr;&ogr; processing module with the converted into a digital signal (12) (A/D) acoustic signal is coupled. 3. Telephone according to claim 2, characterized in that means for compression (K) and/or decompression (D) of the acoustic signal are provided in the module. 4. Telephone according to claim 3, characterized in that the means for compressing (K) the acoustic signal are designed such that the storage requirement is reduced. 5. Telephone according to one of the preceding claims, characterized in that the processing module is coupled to the analog signal (4). 6. Telephone according to one of the preceding claims, characterized in that means (F) for selectively changing the amplitude of different frequency ranges of the acoustic signal are provided in the module. 7. Telephone according to claim 6, characterized in that the means (F) for amplitude processing of the acoustic signal are designed in such a way that a linearization of the transmission frequency response is achieved for the largest possible frequency spectrum of the original signal. 8. Telephone according to one of the preceding claims, characterized in that means for a frequency shift of the acoustic signal are provided in the module. *5 9. Telephone according to claim 8, characterized in that the means for frequency shifting of the acoustic signal are designed in such a way that the largest possible frequency spectrum of an original signal is shifted into the frequency range where the entire reproduction system has a maximum amplification and/or the human ear is most sensitive to &iacgr;&ogr; is the most sensitive. 10. Telephone according to one of the preceding claims, characterized by an analogue memory (M) into which the possibly processed acoustic signal is written. 11. Telephone according to one of the preceding claims, characterized by means (T) for selecting one or more different modifications of the original signal, for example by selecting from a menu displayed on a display element (A) of the telephone. 12. Telephone according to one of the preceding claims, characterized by an amplifier (7, 20) which acts directly on the signal read out from the memory element (M) and optionally modified, e.g. decompressed, and is coupled on the output side to an electroacoustic transducer (8). 13. Telephone according to one of the preceding claims, characterized by a smoothing element (G) for the amplified (20) signal (21), which is arranged in or in front of the electroacoustic transducer (8).