Abstract
The invention relates to an apparatus (100) having an input (11, 12, 15, 16) and an output (13, 14, 17, 18) and having an effect device with volume-regulated audio signals from an audio file or audio source, wherein the audio signals have input levels at an input of the effect device and output levels at an output of the effect device. In order to achieve background sound which is as optimal as possible, the invention proposes that each input level is assigned to a preset output level irrespective of the content of the audio file or audio source, wherein the input level is regulated to the output level.
Claims
1. A device, comprising at least one input, at least one output, and an effects unit with volume-regulated audio signals of an audio file or audio source, wherein the audio signals have input levels at an input of the effects unit and output levels at an output of the effects unit, wherein irrespective of the content of the audio file or audio source, each input level is assigned to a preset output level, wherein the input level is regulated to the output level.
2. The device as claimed in claim 1, wherein the effects unit is an audio compressor.
3. The device as claimed in claim 1, wherein the output level is a value of a non-linear characteristic curve obtained by measured values.
4. The device as claimed in claim 1, wherein the output level is a value of a linear characteristic curve obtained by measured values.
5. The device as claimed in claim 1, further comprising a leveler arranged between the at least one input and the effects unit, wherein the leveler is connected to the at least one input and the effects unit.
6. The device as claimed in claim 1, further comprising a limiter is arranged between the at least one output and the effects unit, wherein the limiter is connected to the at least one output and the effects unit.
7. The device as claimed in claim 6, further comprising a spatializer arranged between the at least one output and the limiter, wherein the spatializer is connected to the at least one output and the limiter.
8. The device as claimed in claim 7, further comprising a filter arranged between the at least one output and the spatializer, wherein the filter is connected to the at least one output and the spatializer.
9. The device as claimed in claim 8, further comprising a leveler is arranged between the at least one output and the filter, wherein the leveler is connected to the at least one output and the filter.
10. The device as claimed in claim 1, wherein the device is in the form of a box-shaped container, USB stick or headphones.
11. The device as claimed in claim 1, wherein the effects unit has a bypass circuit.
12. The device as claimed in claim 1, further comprising a sensor connected to the effects unit, and a control apparatus assigned to the effects unit that controls the effects unit and processes data and/or signals of the sensor.
13. The device as claimed in claim 12, wherein the sensor is connected to an effects unit circuit which produces the volume-regulated audio signals.
14. The device as claimed in claim 12, wherein the effects unit has a crossfader, wherein the control apparatus is assigned to the crossfader, controls the crossfader, and processes data and/or signals of the sensor.
15. An audio arrangement, comprising an audio source, an acoustic system, and a device, the device comprising, at least one input, at least one output, and an effects unit with volume-regulated audio signals of an audio file or audio source, wherein the audio signals have input levels at an input of the effects unit and output levels at an output of the effects unit, wherein irrespective of the content of the audio file or audio source, each input level is assigned to a preset output level, wherein the input level is regulated to the output level, wherein the device is arranged between, and connected to, the audio source and the acoustic system.
16. The audio arrangement as claimed in claim 15, further comprising a sensor is connected to the effects unit, and a control apparatus assigned to the effects unit that controls the effects unit and processes data and/or signals of the sensor.
17. The audio arrangement as claimed in claim 16, wherein the sensor is connected to an effects unit circuit which produces the volume-regulated audio signals.
18. The audio arrangement as claimed in claim 16, wherein the sensor is connected to a crossfader, wherein the control apparatus is assigned to the crossfader, controls the crossfader, and processes data and/or signals of the sensor.
19. A method for improving sound in a room, comprising regulating input levels of an audio signal of an audio file or an audio source in an effects unit to an output level, wherein irrespective of the content of the audio file or audio source, an input level of the audio signal is assigned to a preset output level of the audio signal, wherein the input level is regulated to the output level.
Description
[0038] The invention will be explained in greater detail below with reference to the drawings, in which, illustrated schematically:
[0039] FIG. 1a shows a device according to the invention;
[0040] FIG. 1b shows the characteristic curve of an audio compressor of a device from FIG. 1;
[0041] FIGS. 2a to 2c show acoustic sources and their provision of sound;
[0042] FIG. 3 shows the temporal profile of inventive volume-regulated audio signals and non-volume-regulated audio signals;
[0043] FIG. 4 shows the front side of the device according to the invention from FIG. 1;
[0044] FIGS. 5a to 5b show connection variants of the device according to the invention from FIG. 1;
[0045] FIG. 6 shows a further connection variant of the device according to the invention from FIG. 1;
[0046] FIG. 7 shows a block diagram of an embodiment of the method according to the invention with an arrangement of further effects units in the device from FIG. 1 and the arrangement with a sensor; and
[0047] FIG. 8 shows the characteristic curve of a limiter from the method according to FIG. 7.
[0048] FIG. 1a shows a device according to the invention, which is provided with the reference sign 100 and is in the form of a portable box-shaped container.
[0049] The device 100 has an effects unit that is integrated into the device 100 and is in the form of an audio compressor 10 in the embodiment shown in FIG. 1.
[0050] As is further evident from FIG. 1a, the rear side of the device 100 has audio inputs 11, 12 and audio outputs 13, 14 for audio signals. The audio inputs 11, 12 and audio outputs 13, 14 are connected to the audio compressor 10 via signal lines 11a, 12a, 13a, 14a that run within the device.
[0051] In the embodiment of the device shown in FIG. 1a, the audio inputs 11, 12 are two asymmetrical RCA connectors that form audio inputs for a left and the right audio channel. The audio outputs 13, 14 are the corresponding asymmetrical audio outputs and are in the form of asymmetrical RCA connectors.
[0052] As is further illustrated in FIG. 1a, on its rear side the device 100 is furthermore provided with symmetrical audio inputs 15, 16 and symmetrical audio outputs 17, 18, which are likewise connected to the device 100 via signal lines running within the device 100, these signal lines not, however, being graphically illustrated in FIG. 1a. The audio inputs and outputs 15, 16, 17, 18 are XLR/6.3 mm jack inputs and XLR/6.3 mm jack outputs. The device 100 has the connector 19 for the power supply. The power supply is configured as a 5 V USB connection in the device 100.
[0053] The audio compressor 10 fulfils the function of a volume regulator. To this end, in the audio compressor 10, the input levels of the audio signals of an audio file that have passed into the audio compressor 10 via the signal lines 11a, 12a are regulated to output levels that pass via the signal lines 13a, 14a to the audio outputs 13, 14 of the device 100, wherein, in the audio compressor 10, irrespective of the content of the audio file, each input level is assigned to a preset output level, wherein the input level is regulated to the output level. For this, the audio compressor 10 is in the form of a dynamics processor with corresponding software, wherein the software in turn contains an algorithm that is processed by the processor. The volumes, i.e. the output levels in the audio compressor 10, are preset in this case and are based on empirical values that in turn are values of a recorded characteristic curve. A characteristic curve of this kind is shown in FIG. 1b.
[0054] FIG. b shows a characteristic curve 24 that is made up of a non-linear range 25 and a linear range 20, wherein the ordinate axis 21 is provided with values of the output levels and the abscissa axis 22 is provided with values of the input levels of audio signals of an audio file.
[0055] The characteristic curve 23 in FIG. 1b shows a characteristic curve of audio signals that are not regulated by the audio compressor 10.
[0056] As is evident from FIG. 1b, in contrast, the presets of the output levels manifesting themselves in the characteristic curve 24 provide a linear volume increase in the lower volume regions, i.e. in the linear range 20, to make quiet contents generally louder and then approach the upper volume regions in a non-linear manner, so that the factor of the volume increase that is becoming ever lower can be cut off as far as almost the maximum level of 0 dB and regulated dynamically. The effect created as a result is shown in FIGS. 2a to 2c.
[0057] In order to achieve a comprehensive provision of sound, according to the prior art the acoustic sources 26 that are shown in FIG. 2a and have a conical emission characteristic are installed in what are known as grids, or else in a manner free of patterns and structures, in a number and of a type depending on the area to be provided with sound. The objective of these installations is both to avoid producing the sound overlaps 30 between the acoustic sources 26, as shown in FIG. 2b, so that the volumes of these do not add together or produce propagation time differences and phase differences that are undesirably audible, and to avoid positioning the acoustic sources 26, as shown in FIG. 2a, too far away from one another so that no acoustic gaps/gaps in sound 27 occur. Nevertheless, a situation arises in which there is a non-comprehensive supply of sound over the area to be provided with sound as a result of too few acoustic sources 26 or too many acoustic sources 26, as a result of which, in turn, the perception of sound varies in intensity at different locations in the area to be provided with sound, which contradicts the ideal of the uniform perception of sound.
[0058] In contrast, the device 100 from FIG. 1 stabilizes the sound of an audio signal over all the frequencies such that the emission behavior of the acoustic sources is intensified. An acoustic source therefore emits a constant acoustic pressure that is perceived by the listener as a relatively large emission radius 31, as is evident from FIG. 2c. As a result of the relatively large emission radius 31 of each acoustic source 26 optimized by the device 100, it is possible for the first time to install fewer acoustic sources 26 to provide sound uniformly over an area.
[0059] The effect of the device 100 shown in FIG. 1 is also evident from FIG. 3, which shows the profile of an audio signal, wherein the amplitude axis is provided with the reference sign 32 and the time axis with the reference sign 33.
[0060] The device 100 from FIG. 1 with the effects unit causes what are known as transients 34 to be retained. Transients 34 are a short percussive sound with a high level at the beginning of a sound event. The transients need not be dependent on the pitch, and they are often non-harmonic. Rather, short sound components that appear percussive are involved here. Transients 34 are important for maintaining the sound quality of the sound event since they are an important sound element, specifically the characteristic beginning of a sound event, for example the beat of a drum, or the plucking sound of a violin or double bass string. The device 100 from FIG. la achieves a high-quality sound result by virtue of said device increasing the volume of the transients 34 that are too quiet together with the rest of the sound event, but not changing the form thereof, which is expressed by the waveform 35 shown in FIG. 3, wherein a range 36 of the audio signals that already have an optimum volume are not processed and therefore distorted, i.e. unprocessed audio signals that are shown in the two lower profiles of the audio signals in FIG. 3, and also the audio signals regulated by the device 100 that are shown in the two upper profiles of the audio signals in FIG. 3 and have the same waveform in a range 36 in which unprocessed audio signals already have an optimum volume.
[0061] FIG. 4 shows the front side of the device 100 from FIG. 1b, having a switch unit 37 that can switch at least one applied audio input signal over to mono or stereo processing. As is further evident from FIG. 4, on the device 100 there is a switch 38 that is a bypass switch that switches the device 100 in an active manner (bypass off) or inactive manner (bypass on) to allow the unprocessed and the processed audio input signals to be compared in real time. The optical indicator 39 indicates whether the device 100 is in bypass-on or bypass-off mode.
[0062] In addition, as is further evident from FIG. 4, there are further optical indicators 40, 41, 42, 43 on the device 100 that indicate possible overdriving of an input level of an audio signal. The input level regulator 44 shown in FIG. 4 is appropriately adapted so that there is no overdriving at the input. The output level regulator 45 regulates the output level of an input level regulated in the effects unit 10 such that there is no overdriving at the input of an acoustic system, which is not shown in FIG. 4. In addition to the wired signal transmission, the device 100 shown in FIG. 4 also has wireless communication that is in the form of a Bluetooth connection 46.
[0063] FIGS. 5a, 5b and 6 show various possibilities for the connection of the device 100 from FIG. 1a to other components of an audio system. FIG. 5a shows the connection variant between a number of audio sources 48 of an arrangement 47 of these audio sources 48 that are connected to the audio inputs 51 of the device 100. FIG. 5b shows the connection variant between the device 100 with a plurality of audio compressors 53 and a number of acoustic systems 49, an arrangement 50 of these acoustic systems with their audio inputs 52.
[0064] The arrangement of a number of compressors 53 of a device 100 from FIG. 1 is evident from FIG. 6, the volume-regulated audio signals of which are mixed together in a mixing unit 55 before they are then guided as a mixed signal into at least one audio input of an acoustic system 49. The mixing unit 55 is a device in which a number A of audio input signals are mixed together onto a number B of audio output signals. Mixing units of this type are particularly common in music studios in which several audio sources, for example musical instruments, are down-mixed to a stereo source, that is to say to two channels. In addition to the pure mixing function of audio sources, modern mixing units also include the possibility of incorporating internal and external effects units in order to refine the sound result during the mixing operation.
[0065] FIG. 7 shows an embodiment of the method of the invention, which begins with an audio input signal 56 from at least one audio source that is not shown in FIG. 7, which signal is applied to the device 100 according to the invention from FIG. 1, being regulated by a leveler 57 in terms of its volume level in such a way that there can be no overdriving within the device 100 on account of different base volumes of audio sources. The leveler 57 can adapt the volume(s) of the applied audio source(s) manually or automatically for further processing by the compressor 10. If the volume level is adapted by the leveler 57, the signal is forwarded to the compressor 10, downstream of which there is also connected a limiter, not shown in FIG. 7. After the audio signal 23 has passed through the limiter, the audio input signal 23 passes into the spatializer 58 and then into a filter and effects unit 59. Here, the sound image can be adapted in terms of its frequency image and if necessary can be provided with effects such as hall and echo effects, for example, so as to intensify the sound of the modified sound events. This filter and effects unit 59 operates manually or automatically and is based on analog or digital technologies. After the filter and effects unit 59, the audio input signal 56 passes through the output leveler 60 connected downstream before it becomes the audio output signal 61.
[0066] Optionally, as is further evident from FIG. 7, an effects unit can be connected to a sensor 63, which is a speed sensor, for example, and is in turn situated outside of the device 100 described in FIG. 1 in the embodiment of the invention shown in FIG. 7. The sensor 63 is connected to the compressor 10 in the embodiment of the invention shown in FIG. 7.
[0067] The sensor 63 captures speeds and changes thereto in a motor vehicle, which is not shown in FIG. 7, in a conventional way and converts the data determined in this case in a conventional way into a signal that is guided into a control apparatus that is also not shown in FIG. 7 and is part of the compressor 10. The captured data of the sensor 63 are processed in a processor of the control apparatus that is embedded in the effects unit circuit in the form of the compressor circuit 64. The processor is configured in terms of programming in such a way that the processor controls the compressor circuit 64 depending on the speeds captured, i.e. the processor runs an algorithm of corresponding software in such a way that, irrespective of the content of an audio file or audio source, each input level at the input of the compressor 10 is assigned to a preset output level at the output of the compressor 10, specifically depending on the speeds captured, wherein the input level is regulated to the output level. In the embodiment of the invention shown in FIG. 7, the volume of the volume-regulated audio signals increases in the compressor 10 with increasing speed measured by the sensor 63.
[0068] Alternatively, as is additionally evident from FIG. 7, the sensor 63 is connected to the crossfader 65 of the compressor 10. The processor also controls the crossfader 65 by virtue of the crossfader 65 as mixing unit being able to apply different volumes to two channels in the compressor 10, i.e. the crossfader 65 fades two audio channels into one another such that the volumes or signal intensities of the two channels can change. At one position of the crossfader 65, the signal intensity of the first channel can be at a ratio of 100 to 0 with respect to the second channel. At the center position of the crossfader 65, both channels are equally loud. The channel in the compressor circuit 64, i.e. the channel that has the stereo signal volume-regulated by the compressor 10, and the bypass channel 66 that is also in the compressor 10 are provided to this end. Both channels have the same input signal that corresponds to the input signal at the input of the compressor. The processor is configured in terms of programming in such a way that the processor controls the crossfader 65 depending on the speeds captured, i.e. the processor runs an algorithm of corresponding software in such a way that, depending on the speed, the volume of the input signal of the compressor 10 rises or falls either in the channel in the compressor circuit 64 or in the unregulated bypass channel 66.
[0069] The limiter from FIG. 7 connected downstream also operates with a characteristic curve 62 that is shown in FIG. 8, wherein the axes in FIG. 8 correspond to the ordinate and abscissa axes from FIG. 1b.
[0070] The limiter has a threshold parameter 63 from which the regulation operation begins. This regulation operation starts to have an effect at a greater level than the compressor 10 connected upstream since the limiter should only cut off the peak levels, but not further influence the sound as such. In comparison to the linear region of the characteristic curve 62, when using a hard knee characteristic, there is a curved region 64 at −7 dB that is guided further with headroom to a level point below the 0 dB limit. The rise and fall times of the limiter (attack, release, hold) are selected such that no pumping occurs but the regulating times are not too long so that the limiter can develop its effect to protect against peak levels that possibly arise.
