Volume-dependent audio compensation

Abstract

A method (100) for real-time processing of an audio stream in an audio speaker arrangement comprising at least one audio speaker is proposed. The method (100) comprises receiving (110) an audio stream and a playback volume associated with the audio stream, and determining (120) a playback sound pressure level of the audio speaker based on the playback volume and pre-stored acoustic transmission data. The pre-stored acoustic transmission data associates the playback volume with the playback sound pressure level of the audio speaker. The method further comprises generating (130) a compensation filter based on amplitude compensation data associated with the determined playback sound pressure level, and filtering the audio stream using the compensation filter (140) and applying a gain factor to the audio stream, thereby generating a compensated audio stream. The gain factor is configured such that a total audio power of the audio stream is maintained regardless of the compensation filter. The compensated audio stream is provided (150) to the audio speaker (21). In addition, an audio speaker arrangement and a computer program product are proposed.

Description

Technical Field

[0001] The present invention relates to audio compensation and, more particularly, to a method and apparatus for compensating an audio stream in an audio speaker arrangement. Background Art

[0002] Since the introduction of portable electronic devices (such as the Walkman) in the late 1970s TM ), the availability and quality of portable audio has continued to improve. With a smartphone, every user of such a device has access to an endless supply of music and other audio content. The audio content will typically be experienced through a pair of headphones or other portable speakers.

[0003] Audio content is designed to be played in a specific way, i.e. the authors and producers of the content have certain ideas about how the listener should perceive the content. The same goes for speakers and headphones. The designers of these products often have a vision for how their products should alter audio content in order to create, for example, a brand-specific sound or feel.

[0004] One solution to this problem is to personalize the listening experience by having the listener take a hearing test and adjusting the audio to compensate for the results of the hearing test. An implementation of this concept is proposed in US2006215844, where the audio signal is filtered according to the listener's hearing profile.

[0005] Prior art focuses on the hearing threshold, which is the lowest sound pressure a listener can hear. However, the audio played is rarely played at a volume that corresponds to the hearing threshold.

[0006] Given the above, there is clearly room for improvement. Summary of the invention

[0007] The present invention aims to provide a new type of audio compensation which is an improvement over the prior art and which obviates or at least mitigates the disadvantages discussed above. More specifically, the present invention aims to provide an audio compensation which is determined without a priori knowledge of the audio signal and which is applicable to streaming audio. These aims are achieved by the techniques set out in the attached independent claims, with preferred embodiments defined in the dependent claims associated therewith.

[0008] According to a first aspect of the present invention, a method for real-time processing of an audio stream in an audio speaker arrangement is provided. The audio speaker arrangement includes at least one audio speaker. The method includes receiving an audio stream and a playback volume associated with the audio stream. Thus, a playback sound pressure level of the audio speaker is determined based on the playback volume and pre-stored acoustic transmission data. The pre-stored acoustic transmission data associates the playback volume with the playback sound pressure level of the audio speaker. The method also generates a compensation filter based on amplitude compensation data associated with the determined playback sound pressure level, filters the audio stream using the compensation filter, and applies a gain factor to the audio stream, thereby generating a compensated audio stream. The gain factor is configured so that the total audio power of the audio stream is maintained regardless of the compensation filter. The compensated audio stream is provided to the audio speaker.

[0009] In one embodiment of the method, the amplitude compensation data is based on equal loudness contours. By basing the amplitude compensation data on equal loudness, the audio stream will be perceived in the same way regardless of the playback sound pressure level.

[0010] In another embodiment, the amplitude compensation data includes amplitude compensation values ​​for a plurality of different frequencies and a plurality of playback volumes.By adding frequency correlation, the audio stream will be perceived in the same way regardless of the playback sound pressure level.

[0011] In an additional embodiment, the method further includes the step of receiving an updated playback volume that is different from the playback volume, and determining an updated playback sound pressure level for the audio speaker based on the updated playback volume and pre-stored acoustic transmission data. An updated compensation filter is generated based on amplitude compensation data associated with the determined updated playback sound pressure level. The audio stream is filtered to generate a compensated audio stream using the updated compensation filter. The compensated audio stream is provided to the audio speaker in sync with the updated playback volume using the updated compensation filter. Enabling the audio stream to be provided in sync with changes in the playback volume allows the volume-compensated audio stream to be delivered to the user without interruption when the playback volume changes.

[0012] In one embodiment, the filtering step further comprises applying a gain factor to the audio stream, thereby generating a compensated audio stream. The gain factor is configured such that the total audio power of the audio stream is maintained regardless of the updated compensation filter. This is useful because the application of the compensation filter may have a decreasing or increasing effect on the audio stream.

[0013] In yet another embodiment, the audio stream is filtered in audio frames. Having the audio stream in frames makes it possible to process, handle and transmit the audio stream in chunks.

[0014] In another embodiment, the steps of determining the updated playback sound pressure level and generating the updated compensation filter are performed during the intermediate time between filtering of two consecutive audio frames. This allows the playback sound pressure level to be changed substantially instantaneously without compromising the compensation of the audio stream.

[0015] In one embodiment, the step of generating the compensation filter is additionally based on a hearing profile associated with a user of the audio speaker arrangement.Processing the audio stream also based on the hearing profile will further enhance the user's listening experience.

[0016] In another embodiment, the audio stream is formed in a channelized audio format. This is useful when enjoying an audio stream comprising more than one channel, such as a stereo audio stream or other multi-channel audio stream.

[0017] In one embodiment, the channelized audio format includes a left channel audio stream and a right channel audio stream. In this embodiment, the step of generating the compensation filter further includes generating a left compensation filter for the left channel audio stream and a right compensation filter for the right channel audio stream. This is beneficial when enjoying a stereo audio stream and / or when the user's hearing profile is different on the left ear and the right ear.

[0018] In one embodiment, the compensation filter is a digital filter with a linear phase response, such as a FIR filter. FIR filters are beneficial because they can have an exact linear phase, they are always stable, they are efficiently implemented in hardware, and the filter startup transient has a finite duration.

[0019] In a second aspect of the invention, an audio speaker arrangement is introduced, comprising: at least one audio speaker; at least one memory; at least one external interface; and at least one controller, which is operably connected to the at least one audio speaker, the at least one memory and the at least one external interface. The at least one controller is configured to: receive an audio stream and a playback volume associated with the audio stream in real time via the external interface; and determine the playback sound pressure level of the audio speaker based on the playback volume and pre-stored acoustic transmission data. The pre-stored acoustic transmission data associates the playback volume with the playback sound pressure level of the audio speaker. This is used to generate a compensation filter based on amplitude compensation data associated with the determined playback sound pressure level, and to filter the audio stream using the compensation filter and apply a gain factor to the audio stream to generate a compensated audio stream. The gain factor is configured so that the total audio power of the audio stream is maintained regardless of the compensation filter. The compensated audio stream is provided to the audio speaker.

[0020] In one embodiment of the audio speaker arrangement, the external interface is a wireless interface.Having a wireless external interface increases the usability, comfort and flexibility of the audio speaker arrangement.

[0021] In one embodiment of the audio speaker arrangement, it is configured to perform the method for processing an audio stream as disclosed above.

[0022] According to a third aspect, a computer program product is provided. The computer program product is configured to cause the controller to receive an audio stream and a playback volume associated with the audio stream in real time when executed by the controller. The computer program product is also configured to determine the playback sound pressure level of the audio speaker based on the playback volume and pre-stored acoustic transmission data. The pre-stored acoustic transmission data associates the playback volume with the playback sound pressure level of the audio speaker. Thus, a compensation filter is generated based on amplitude compensation data associated with the determined playback sound pressure level, and the audio stream is filtered using the compensation filter, and a gain factor is applied to the audio stream, thereby generating a compensated audio stream. The gain factor is configured so that the total audio power of the audio stream is maintained. The compensated audio stream is provided to the audio speaker.

[0023] In one embodiment, the computer program product is further configured to cause the controller to perform the method for processing an audio stream as disclosed above. BRIEF DESCRIPTION OF THE DRAWINGS

[0024] Embodiments of the invention will be described hereinafter; with reference to the illustrated drawings which illustrate non-limiting examples of how the inventive concept can be reduced to practice.

[0025] Figure 1a is a diagram of an audio playback system according to an embodiment of the present invention.

[0026] Figure 1b is a block diagram of a signal flow of an audio playback system according to an embodiment of the present invention.

[0027] Figure 2 is a graph of equal loudness lines according to an embodiment of the present invention.

[0028] Figure 3 is a graph of the increment between two equal loudness lines according to an embodiment of the present invention.

[0029] Figure 4 FIG. 1 is a block diagram showing the principle of compensating audio streams according to an embodiment of the present invention.

[0030] Figures 5a to 5c Detailed Description of the Invention Different views of audio streams and their associated times according to an embodiment of the present invention.

[0031] Figure 6 A schematic block diagram of pre-stored acoustic transmission data for explanation according to an embodiment of the present invention.

[0032] Figure 7 is a diagram of an audio speaker arrangement in use according to an embodiment of the present invention.

[0033] Figures 8a to 8b is an exemplary graph of a compensation filter according to an embodiment of the present invention.

[0034] Fig. 9 FIG. 1 is a schematic diagram of the generation of a compensation filter according to an embodiment of the present invention.

[0035] Fig.10 is a schematic view of an audio speaker arrangement according to an embodiment of the present invention.

[0036] Fig.11 is a schematic diagram of audio processing according to an embodiment of the present invention.

[0037] Figures 12a to 12b It is a schematic block diagram of a method for processing an audio stream according to an embodiment of the present invention.

[0038] Fig.13 Schematic diagram of a computer program product according to an embodiment of the present invention.

[0039] Fig.14 is a diagram of an audio speaker arrangement according to an embodiment of the present invention.

[0040] Fig.15a is a graph of a compensation filter according to an embodiment of the present invention.

[0041] Fig.15b is a graph of a hearing profile according to an embodiment of the present invention. DETAILED DESCRIPTION

[0042] Hereinafter, specific embodiments will be more fully described with reference to the accompanying drawings. However, the present invention may be embodied in many different forms and should not be construed as limited to the embodiments set forth herein; rather, these embodiments are provided by way of example so that this disclosure will be thorough and complete and will fully convey the scope of the invention, as defined in the appended claims, to those skilled in the art.

[0043] Herein, the word playback system, playback device or playback volume means playback in the sense of reproducing the received audio data. The term does not mean playback in the restricted form of playback of stored audio or media, but covers all forms of media, such as real-time media, stored media, streaming media, progressive download media, etc. In addition, the term compensation means compensation in its broadest form and can be considered synonymous with the term processing in the present disclosure. This applies not only to describing actions, but also to describing objects, such as compensation filters.

[0044] exist Figure 1a In FIG. 1 , an audio playback system 1 is shown. The system 1 comprises an electronic device 10 which may be configured to transmit audio to an audio speaker arrangement 20 via an external interface 30. The audio speaker arrangement 20 comprises an electronic device 10 which may be configured to provide audio output to a user 40 (e.g., Figure 7 Typically, the audio speaker arrangement 20 includes two audio speakers 21 to achieve stereo playback of the audio output.

[0045] The audio speaker arrangement 20 is in one embodiment a pair of headphones 20, and in another embodiment a pair of wireless headphones 20. The electronic device 10 may be a portable electronic device 10, such as a cellular phone, a portable music player, a tablet computer, a laptop computer, or the like.

[0046] The electronic device 10 transmits audio to the speaker arrangement 20 in a form suitable for the external interface 30 and the audio speaker arrangement 20. The transmission may be digital or analog, which in the concept of the present disclosure means the way of presenting audio information. The external interface 30 may be a wired or wireless external 30 interface. If the external interface 30 is a wired interface, the transmission of audio from the electronic device 10 may be in analog form or digital form depending on the speaker arrangement 20. If the external interface 30 is a wireless interface, the transmission of audio from the electronic device 10 will generally be in digital form. Although most of the description will be related to the external interface 30 in the form of a wireless digital interface, the details of the present disclosure are applicable regardless of the type of the external interface 30. After reading the present disclosure, those skilled in the art will understand how to change the present invention to make it also successful with a wired analog or digital external interface 30.

[0047] Figure 1a The audio playback system 1 introduced in is used to provide an audio stream 33 to a user 40 at a specific volume. Figure 1b, some definitions will be used to explain the flow of the audio stream 33. The audio stream 33 is recorded at a certain moment at a level called the master level 12. This master level 12 is basically the gain of the initial recording. The master level 12 is roughly the same between different recording media, which makes it possible to listen to different media streams without adjusting the playback volume 15.

[0048] The audio stream 33 is provided to the electronic device 10 at its master level 12. The audio stream 33 may be provided to the electronic device 10, for example, as the audio component of streaming media from a streaming service, retrieved from a storage unit of the electronic device 10, etc. When listening to the audio stream 33, a user 40 of the electronic device 10 will set a playback volume 15 for the audio stream 33. This setting of the playback volume 15 may be performed by adjusting the playback volume 15 with a button or control on the electronic device 10 or the audio speaker arrangement 20.

[0049] The audio stream 33 is provided from the electronic device 10 to the audio speaker arrangement 20 via the external interface 30. Since the audio stream 33 is usually provided at the master level 12, the audio stream 33 is provided together with the playback volume 15. It is possible for the electronic device 10 to change the master level 12 of the audio stream 33 before providing the audio stream 33 to the audio speaker arrangement 20. The audio speaker arrangement 20 receives the audio stream 33 and the playback volume 15, and uses both to provide the audio stream at a playback sound pressure level 52 to the user 40 of the audio speaker arrangement 20. The playback sound pressure level 52 is the sound pressure level at which the user 40 receives the audio signal 33. The playback volume 15 will result in the playback sound pressure level 52 provided by the audio speaker 21. During the playback of the audio stream 33, the playback volume 15 can be changed by a first factor, and this results in the playback sound pressure level 52 being changed by a second factor. However, the playback sound pressure level 52 will depend on the type of the audio speaker arrangement 20. The association between the first factor and the second factor can be any type of linear or non-linear association. This correlation is usually unknown to the electronic device 10, since in some cases one electronic device 10 may be used with many different audio speaker arrangements 20. When an audio signal is mastered, the playback sound pressure level 52 is important if the audio signal should be perceived as intended.

[0050] The audio signal includes a frequency spectrum, and this frequency spectrum of the audio signal will be perceived differently at different playback sound pressure levels 52. When comparing the equal loudness lines 300 at different sound pressure levels (see Figure 2 One problem solved by the inventive concepts of the present disclosure is to enable the audio stream 33 to be perceived as intended, i.e., as it would be perceived when mastered at the master level 12, regardless of the playback sound pressure level 52.

[0051] Figure 2A plurality of equal loudness lines 300 are shown in FIG. The equal loudness lines 300 are a measure of sound pressure level, usually expressed in decibels as dB sound pressure level SPL. The sound pressure level is provided by a spectrum, usually the audible spectrum of humans. Each point in the equal loudness lines 300 represents a sound pressure level whose constant loudness is perceived by the listener when compared to a pure tone presented at a particular phon at 1 kHz. Phon is defined in DIN 45631 and ISO532, and will be defined herein to mean that a 1 kHz tone with 50 dB SPL has a loudness of 50 phons, and all other frequencies perceived at the same loudness as this signal also have a loudness of 50 phons.

[0052] exist Figure 2 In , draw equal loudness 300 lines for different parties. Figure 2 , equal loudness lines 300 are presented for 0, 20, 40, 60, 80, and 100 phons. Figure 2 The loudness contours 300 are the loudness contours 300 according to ISO 226:213. These contours 300 are corrected compared to the Fletcher-Munson curves provided by Harvey Fletcher and Wilden A. Munson in 1933. The Fletcher-Munson curves are often referred to as the first description of the concept of equal loudness. If the sound pressure level has no effect on how the listener perceives the frequency spectrum of the audio signal, then Figure 2 The offset between the lines 300 in will be the same at all frequencies.

[0053] exist Figure 3 , the delta 310 between the equal loudness lines 300 of 20 phons and 40 phons is shown. As can be seen, the sound pressure level difference 310 at 1 kHz is 20 dB as expected, as shown by the dashed line, but differs greatly from the expected 20 dB above and below 1 kHz. This effect causes the audio playback to be perceived as having different frequency spectra depending on the sound pressure level of the playback.

[0054] The inventors of the present disclosure have realized, after careful reasoning and creative thinking, that the effects described above can be compensated for in the playback system 1 when the audio stream 33 is played from the electronic device 10. The solution proposed herein will compensate the audio stream 33 based on the selected playback volume 15, and will change the compensation substantially instantly when the playback volume 15 changes.

[0055] refer to Figure 4 , the principle of compensating the audio stream 33 according to the present invention will be illustrated. Figure 1aThe playback system 1 is introduced to explain these principles, and many of the processing steps are performed in what is referenced as the audio speaker arrangement 20. However, it should be emphasized that the processing steps may well be performed in the electronic device 10, or distributed between the electronic device 10 and the audio speaker arrangement 20.

[0056] exist Figure 4 In the schematic block diagram of , the electronic device 10 communicates a signal 31 via an external interface 30. The signal 31 includes an audio stream 33 and control data 32 associated with the audio stream 33. The audio speaker arrangement 20 receives the signal 31 via the external interface 30 using a transceiver module 22 suitable for the external interface 30. For example, if in one embodiment the external interface is a Bluetooth low energy interface, the transceiver module 22 is a Bluetooth low energy transceiver module, and if in one embodiment the external interface is a wired serial interface, the transceiver module 22 is a wired serial transceiver module. The transceiver module 22 receives the signal 31, analyzes the signal 31 and detects its components, such as the audio stream 33 and its associated control data 32. Note that the detection and analysis can be performed by any suitable control circuit, and this function can be included in other modules of the audio speaker arrangement 20. The control data 32 is communicated to the control module 24, and the audio stream 33 is communicated to the digital signal processing DSP module 28.

[0057] The control data 32 includes a playback volume 15 associated with the audio stream 33, and the control module 24 is provided with the control data 32 and pre-stored acoustic transmission data 50 (shown in FIG. Figure 6 The pre-stored acoustic transmission data 50 describes how the playback volume 15 is converted into a playback sound pressure level 52 of the audio speaker 21. The pre-stored acoustic transmission data 50 will be further explained in the following section. The control module 24 generates a compensation filter 320 (such as Fig. 9 ). The compensation filter 320 is represented in one embodiment as digital filter coefficients of a finite impulse response FIR filter. The compensation filter 320 may be communicated to the DSP module 28 as a DSP control signal 25. The compensation filter 320 may be any type of suitable filter or equalization function, such as an infinite impulse response IIR filter.

[0058] The DSP module 28 includes functionality for applying the digital filter coefficients provided by the control module 24 when compensating the audio stream 33 so as to provide the compensated audio stream 27 to the audio speaker 21. In addition to applying the compensation filter 320 to the audio stream 33, the DSP module 26 may also be configured to apply other processing functions to the audio stream, such as noise cancellation, sound coloration, audio and music effects, bass enhancement, stereo widening, 3D sound effects, etc.

[0059] Before the compensated audio stream 27 is fed to the audio speaker 21, it will typically undergo a digital-to-analog D / A module 28, which converts the digital compensated audio stream 27 into an analog audio signal 29 that is provided to the audio speaker 21. The analog audio signal 29 is a time-varying audio signal having a voltage amplitude that is related to the playback volume 15. The voltage amplitude of the analog audio signal 29 determines the playback sound pressure level 52 of the audio speaker 21.

[0060] As mentioned previously, there are different implementations in which the playback sound pressure level 52 of the audio speaker arrangement 20 may be controlled. Depending on how the playback sound pressure level 52 is controlled, handling of changes to the playback volume 15 by the user 40 of the electronic device 10 may be handled differently.

[0061] In one embodiment, the playback sound pressure 52 level is controlled by applying a digital gain to the audio stream 33 during processing by the DSP module 26. The DSP module 26 will perform the change in playback volume 15, and the application of the updated compensation filter 320 can be synchronized with the application of the updated playback volume 15'. The synchronization can be controlled by the control module 24 or the DSP module 26.

[0062] In another embodiment, the playback sound pressure level is controlled by applying a digital gain and / or an analog gain to the compensated audio stream 28 during processing by the A / D module 28. In such an embodiment, the updated gain is communicated by the control module 24 to the A / D module 28, which Figure 4 , and the updated compensation filter 320 is communicated to the DSP module 26. In this embodiment, the control module 24 is responsible for synchronizing the application of the updated compensation filter 320 with the updated playback volume 15'.

[0063] The embodiments regarding the updating of the playback volume 15 and the synchronization of the updated playback sound pressure level 52 with the updated compensation filter 320 are considered as non-limiting examples. The embodiments described may be combined in the case where the volume change results in a change in the digital gain of the DSP module 26 and a change in the analog gain of the A / D module 28. It is also possible that the digital gain is controlled by both the DSP module 26 and the A / D module 28.

[0064] refer to Figures 5a to 5c , the processing of signal 31 will be described in further detail. Figure 5a , an exemplary audio stream 33 is shown as included in the signal 31 communicated via the external interface 30. The audio stream 33 is shown as a time-varying signal for illustrative purposes only. The audio stream 33 is typically formed into data packets and transmitted on a channel such as that used for audio streaming in the external interface 30.

[0065] exist Figure 5bIn the embodiment, the transceiver module 22 receives an audio stream 33 in an audio frame 23. Each audio frame 23 stores data representing the audio of the frame time TF. In one embodiment, each audio frame 23 stores 512 audio data samples sampled at 44.1kHz, which corresponds to a frame time TF of about 11.6ms. It should be noted that the frame size may be different between systems. The audio frame 23 is usually received via the external interface 30 at a rate that depends on the transmission rate of the external interface 30. The problem of undesired interruption of the playback of the audio stream 33 associated with the external interface 30 can be alleviated by an audio speaker arrangement 20 with a frame buffer. The frame buffer is configured to buffer the frame 23 and reduce the risk of jitter in the playback. Assuming that the frame 23 is received at a rate substantially equal to the frame time TF, only one erroneously received frame 23 can cause jitter in the playback of the audio stream 33.

[0066] The audio frame 23 is provided to the DSP module 22, where it is processed by, for example, a compensation filter 320. The processing of each frame 23 by the DSP module 22 is referred to as a processing time TP, and the processing time TP is shorter than the frame time TF. The time difference between the frame time TF and the processing time TP is an idle time T1. During the idle time T1, the DSP module will typically be in a sleep mode in order to save power, and this time T1 can be used to apply and / or generate and / or update the compensation filter 320. The generation of the compensation filter 320 is suitable for implementations in which, for example, the control module 24 and the DSP module 26 are the same module 24, 26 or the control module 24 instructs the DSP module 26 to generate the compensation filter 320 based on parameters from the control module 24. The filtered audio frame 23 is provided to the D / A module 28 as a compensation audio stream 27. The compensation audio stream 27 will typically be buffered in an output buffer in a manner similar to that of the previously disclosed frame buffer. The output buffer is typically clocked by a D / A clock, and the output is provided to the audio speaker 21. In the above described embodiments, the D / A clock is typically associated with the sampling rate of the audio data, 44.1 kHz. The output buffer is a system design parameter and in one embodiment is configured to hold two audio frames 23 .

[0067] By utilizing the idle time T1 between two consecutive audio frames 23 to generate the compensation filter 320, it is possible to seamlessly and substantially instantly update the compensation filter 320 in sync with changes in the playback sound pressure level 52 of the audio speaker 21 during playback of the audio stream 33. The teachings presented herein enable real-time changes in the playback sound pressure level 52 of the audio stream 33 from the perspective of the user 40. In other words, the time it takes from the audio speaker receiving the updated playback volume 15 to the arrangement 20 to the change in the playback sound pressure level 52 is very short, in the range of 10ms.

[0068] refer to Figure 6, the pre-stored acoustic transmission data 50 will be explained in further detail. The playback volume 15 is typically provided by the electronic device 10 and can be in many different forms. The playback volume 15 can be a percentage, where 100% is the highest possible playback volume, or it can be discrete steps from, for example, 0 to 15, where 15 corresponds to the highest playback volume 15. The playback volume 15 is configurable in either case, and it must be related to the playback sound pressure level 52 of the audio speaker 21. The playback volume 15 is not necessarily directly related to the playback volume of the audio speaker arrangement 20. The audio speaker arrangement 20 can be configured to have more or fewer steps of playback volume than the electronic device 10. The pre-stored acoustic transmission data 50 is used at this time.

[0069] The pre-stored acoustic transmission data 50 provides a means for correlating the playback volume 15 of the electronic device 10 with the playback sound pressure level 52 of the audio speaker arrangement 20. As an example, if the audio speaker arrangement 20 is provided with functionality allowing 32 volume steps (0 to 31), then each of these volume steps is correlated with a playback sound pressure level 52. If the electronic device 10 is provided with functionality allowing 16 volume steps (0 to 15), then selecting an intermediate volume step, i.e., step 7, on the electronic device 10 will result in selecting an intermediate volume step, i.e., step 15, in the audio speaker arrangement 20.

[0070] The pre-stored acoustic transmission data 50 may be formed in many different ways. It may be formed as a lookup table with absolute values ​​of playback sound pressure levels 52 associated with different volume steps of the audio speaker arrangement 20. Alternatively, one volume step of the audio speaker arrangement 20 may be selected as a baseline volume step associated with an absolute value of the playback sound pressure level 52, and other volume steps of the audio speaker arrangement may be associated with playback sound pressure levels 52 that are increments of the absolute value of the baseline volume step.

[0071] In one embodiment, the pre-stored acoustic transmission data 50 includes one or more playback volumes 15 , each of which is associated with a corresponding playback sound pressure level 52 .

[0072] Optionally, the pre-stored acoustic transmission data 50 may include more than one playback sound pressure level 52 for each playback volume 15. The multiple playback sound pressure levels 52 may be playback sound pressure levels for different playback frequencies. In one embodiment, one playback sound pressure level 52 is associated with each playback volume 15 of the audio speaker arrangement 20, and the playback sound pressure level 52 is a playback sound pressure level 52 determined with a 1 kHz test tone. In another embodiment, one playback sound pressure level 52 is associated with each playback volume 15 of the audio speaker arrangement 20, and the playback sound pressure level 52 is a playback sound pressure level 52 determined as an average sound pressure when the audio speaker 21 is subjected to white noise.

[0073] In some of the embodiments, the pre-stored acoustic transmission data 50 is arranged in a table.

[0074] Not all possible playback volumes 15 need to be listed in the pre-stored acoustic transmission data 50. If a playback volume 15 not listed in the pre-stored acoustic transmission data 50 is selected, the relevant playback sound pressure level 52 may be calculated by interpolation of the playback volumes 15 and their associated playback sound pressure levels 52 immediately above and below the unlisted playback volume 15. The pre-stored acoustic transmission data 50 may be stored so that the playback volumes are associated with equal loudness contours 300 for certain parties.

[0075] In some embodiments, the pre-stored acoustic transmission data 50 is represented as an equation describing a mathematical relationship between the playback volume 15 and the playback sound pressure level 52. The pre-stored acoustic transmission data 50 is typically specific to one model of the audio speaker arrangement 20. The pre-stored acoustic transmission data 50 may be specific to a series of models of the audio speaker arrangement 20. The pre-stored acoustic transmission data 50 may be specific to each unit of the audio speaker arrangement 20. The pre-stored acoustic transmission data 50 may be stored in the audio speaker arrangement 20 when the audio speaker arrangement 20 is created, or provided to the audio speaker arrangement later via a firmware upgrade or, for example, when a user 40 of the audio speaker arrangement 20 decides to start using the volume-related compensation of the present disclosure.

[0076] The generation of the pre-stored acoustic transmission data 50 may be carried out by recording the playback sound pressure level 52 at a particular playback volume 15 available to the audio speaker arrangement 20 .

[0077] refer to Figure 7 , an embodiment in which the audio speaker arrangement 20 is a pair of earphones 20 will be used to explain the generation of pre-stored acoustic transmission data 50. The earphones 20 are worn during use by the user 40 so that the audio from the earphones 20 is directed substantially directly to the ears of the user 40. The playback sound pressure level 52 that will be used to compensate the audio stream 33 is the sound pressure level 52 at the ears of the user 40. This means that the recording is preferably done at the area 41 in or around the ears of the user 40. One possibility is to use a simulated human head instead of the user 40. Simulated human heads including recording means (e.g. microphones) are readily available on the market.

[0078] Using the details of the previous section, it is possible to compensate the audio stream 33 based on the playback volume 15. As described above, the sound (e.g., audio) is perceived differently depending on the playback sound pressure level 52, and different compensation filters 320 can be used for different playback sound pressure levels 52. Figure 3 The difference between the equal loudness contours 300 of 20 phons and 40 phons presented in FIG. 3 can be obtained by Figure 8a. Applying this compensation 320 when changing from a 1 kHz playback sound pressure level of 40 dB SPL to a 1 kHz playback sound pressure level of 20 dB SPL will compensate for the difference in perceived sound between the 40 phon equal loudness line 300 and the 20 phon equal loudness line 300. In this example, the compensation filter 320 is intended to be applied in conjunction with a 20 dB reduction in the playback sound pressure level 52. Figure 8a This means that the sound pressure level 52 will be reduced by 20 dB at 1 kHz, but only by about 10 dB at 20 Hz and even below 20 kHz. By having pre-stored acoustic transmission data 50, a compensation filter 320 can be generated based on the playback volume 15 so that the perceived sound pressure level is substantially the same for all frequencies, regardless of the playback sound pressure level 52.

[0079] like Figure 8b As shown in , the compensation filter 320 can be represented as amplitude compensation data 330 including one or more amplitude compensation values ​​335. The amplitude compensation value 335 describes the compensation per frequency. There may be any number of amplitude compensation values ​​335, each value being associated with a frequency or frequency range.

[0080] refer to Fig. 9 , the compensation filter 320 will be generated based on the playback sound pressure level 52 as determined from the pre-stored acoustic transmission data 50 and the playback volume 15. The compensation filter 320 may be generated by applying the amplitude compensation data 330. Figure 8b , the amplitude compensation data 330 includes the differences in amplitude filtering for a plurality of different frequencies and playback sound pressure levels 52. These differences may be represented by amplitude compensation values ​​335. The amplitude compensation data 330 may include a reference level at a reference playback sound pressure level. At the reference level, the difference in amplitude filtering is zero, and other playback sound pressure levels 52 in the amplitude compensation data 330 are described relative to the reference level. If there is a lack of a match to a playback sound pressure level 52 in the amplitude compensation data 330, then interpolation may be performed between adjacent playback sound pressure levels 52 and their associated amplitude compensation data 330. For pre-stored acoustic transmission data 50, the amplitude compensation data 330 may be a mathematical function that converts a given playback sound pressure level 52 into a compensation filter 320.

[0081] In one embodiment, the amplitude compensation data 330 is based on the equal loudness contours 300. In one embodiment, the amplitude compensation data 330 has a reference level corresponding to the playback sound pressure level 52 of 60 dB SPL. In one embodiment, the amplitude compensation data 330 has a reference level corresponding to the equal loudness contours 300 of 60 dB SPL.

[0082] The compensated audio stream 27 will have a different frequency content than the uncompensated audio stream 33. High frequencies may be amplified more than low frequencies, and vice versa. Depending on the frequency content of the audio stream 33, it may be that when the compensation filter 320 is applied, the playback sound pressure level 52 may be considered lower. For example, assume that the audio stream 33 is played back with the compensation filter 320 applied. If the playback volume 15 is increased and an updated compensation filter 320' is determined, the updated compensation filter may result in a substantially unchanged or even lower playback sound pressure level 52. Further assume that the playback volume 15 is intended to increase the playback sound pressure level 52 by 5dB. If an updated compensation filter is applied that provides 5dB attenuation at low frequencies and 0dB attenuation at high frequencies, the playback sound pressure level will increase by 5dB at high frequencies and 0dB at low frequencies. Now, if most of the audio power of the audio stream 33 is in low frequencies, i.e., the bass is very heavy, then with the updated compensation filter 320' applied, the cumulative audio power at all frequencies will be substantially reduced.

[0083] In order to alleviate the above problems, the inventors have realized that the signal frequency content of the average audio stream can be estimated at the main level 12 and the average audio power of the average audio stream can be estimated. The frequency content of the average audio stream is represented by a typical frequency distribution, and this can be provided by analyzing a large number of audio streams in order to calculate the average audio stream and its associated typical frequency distribution. From this, a gain factor can be generated that ensures that the total audio power of the average audio stream is substantially the same with and without the application of the compensation filter 320. The gain factor can be applied to the audio stream 33 together with the compensation filter 320, or applied to the audio stream 33 at other stages of the processing of the audio stream 33.

[0084] In addition to compensating the audio stream 33 based on the playback volume 15, the audio stream may, in one embodiment, be compensated relative to the hearing profile 340 of the user 40. Fig.11 Schematically shown in FIG, wherein a compensation filter 320 is generated based on amplitude compensation data 330 and a hearing profile 340 of a user 40. The hearing profile 340 may be any kind of suitable hearing profile 340, and may include a separate hearing profile 340 for each ear of the user 40. The hearing profile 340 may be provided to the audio speaker arrangement 20 from the electronic device 10. The electronic device 10 may be configured to generate the hearing profile 340 based on a hearing test of the user 40, or to obtain the hearing profile 340 from a hearing profile repository. Such a repository may be, for example, a database hosted on a cloud server. The hearing profile 340 may well suggest different compensation for each ear of the user 40.

[0085] Many audio speaker arrangements 20 include more than one audio speaker 21 and provide for playback of an audio stream 33 having more than one audio channel. Fig.10 As explained, the invention is also applicable to multi-channel audio systems.

[0086] exist Fig.10 , a two-channel audio speaker arrangement 20 is shown. If the signal 31 received by the transceiver module 22 includes an audio stream 33 having more than one channel, then these channels can be identified, separated, and provided to each of the audio speakers 21L, 21R of the audio speaker arrangement 20. Typically, the audio stream is a stereo audio stream including a left channel audio stream 33L and a right channel audio stream 33R. The left channel audio stream 33L is intended for the left audio speaker 21L, and the right channel audio stream 33R is intended for the right audio speaker 21R.

[0087] exist Fig.10 , the transceiver module 22 is shown as identifying and providing the audio streams 33L, 33R to the left DSP module 26L and the right DSP module 26R, respectively, which identification may well be done in, for example, the DSP module 26, but is done in the transceiver module 22 for illustrative purposes. Each of the audio streams 33L, 33R may be subjected to a different compensation filter 320, one reason may be that the balance between the channels is different, resulting in the left playback volume and the right playback volume being different from each other. Another reason may be that the audio speakers 21L, 21R have different characteristics and therefore have different pre-stored acoustic transmission data 50. This means that when equal audio streams 33 are provided to the respective audio speakers 52L, 52R, the left sound pressure level 52L will be different from the right sound pressure level 52R. As explained in the previous section related to the hearing profile 340 of the user 40, there are also reasons for differences related to the user 40 of the audio speaker arrangement.

[0088] refer to Fig.10 The description given is made with respect to dual speakers 21L, 21R and a stereo stream, the same reasoning applies to an audio speaker arrangement 20 having even more audio speakers 21. This may be the case, for example, in a car stereo system or a surround system. The same reasoning applies if the audio stream 33 is a mono audio stream 33 provided to more than one audio speaker 21, 21L, 21R.

[0089] Now refer to Fig.12a A method 100 for processing an audio stream 33 is presented. The method 100 is presented without all possible detailed steps regarding the details of the method 100, as these details are presented in other parts of the present disclosure. It is obvious to those skilled in the art that they can also be implemented on the method 100 for compensating the audio stream 33.

[0090] The method 100 is initiated by receiving 110 a signal 31 comprising an audio stream 33 and control data 32. The control data comprises at least a playback volume 15 associated with the audio stream 33. As mentioned above, the audio stream 33 may optionally be received in audio frames 23. These audio frames 23 may be buffered in a receiving buffer.

[0091] Based on the playback volume 15, the method 100 proceeds by determining 120 a playback sound pressure level 52 using pre-stored acoustic transmission data 50 as previously presented. As previously mentioned, there are embodiments in which a plurality of playback sound pressure levels 52 may be determined based on, for example, the user 40, the signal 31 and / or the audio speaker arrangement 20.

[0092] The determined playback sound pressure level 52 is used together with the amplitude compensation data 330 to generate 130 the compensation filter 320. The generation 130 of the compensation filter 320 may also be based on the hearing profile 340 of the user 40. The amplitude compensation data 330 may include one or more amplitude compensation values ​​335 based on the equal loudness contours 300.

[0093] The generated 130 compensation filter 320 is used to filter 140 the audio stream 33 in order to provide 150 the compensated audio stream 27 to the audio speaker 21 .

[0094] The proposed method 100 allows processing of an audio stream based on the playback volume 15. As presented in the previous section, the user 40 can change the playback volume 15 without annoying delays in changes in the playback sound pressure 52 or delays in updating or changing the compensation filter 320. The method 100 for updating the compensation filter is substantially the same as the method 100 for compensating the audio stream 33. Figure 12b , receiving an updated playback volume 15′, which is used to determine 120′ an updated playback sound pressure level 52 in the same manner as the playback sound pressure level 52 was determined 120. An updated compensation filter 320 is generated 130′ similar to the generation 130 of the compensation filter 320. The updated compensation filter 320 is applied to the audio stream as disclosed in the previous part of the present disclosure and is used to filter 40 the audio stream 33 and provide 150 the compensated audio stream 27 to the audio speaker 21.

[0095] The method 100 is preferably performed by the audio speaker arrangement 20 , but may in embodiments be performed solely or partially by the electronic device 10 .

[0096] exist Fig.13 As shown in FIG. 1 , the method 100 may be implemented as a computer program product 600 , such that when the computer program product 600 is executed by a device such as a controller, a computer, a DSP, etc., the device performs the method 100 as disclosed herein.

[0097] In one embodiment of the computer program product 600, it is configured to, when executed by the controller 200, cause the controller 200 to control the transceiver module 22 to receive the audio stream 33 and the playback volume 15 associated with the audio stream 33 through the external interface 30. The computer program product 600 is further configured to control the control module 24 to determine the playback sound pressure level 52 of the audio speaker arrangement 20 based on the playback volume 15 retrieved from the memory 700 and the pre-stored acoustic transmission data 50, so that the playback volume 15 is associated with the playback sound pressure level 52 of the audio speaker 21. In addition, the computer program product 600 is configured to control the control module 24 to generate the compensation filter 320 based on the amplitude compensation data 330 associated with the determined playback sound pressure level 52 retrieved from the memory 700, and control the DSP module 26 to filter the audio stream 33 using the compensation filter 320 to generate the compensated audio stream 27. The computer program product 600 is further configured to control the D / A module 28 to provide the compensated audio stream 27 to the audio speaker 21 .

[0098] exist Fig.14 , a block diagram of an audio speaker arrangement 20 is shown. The audio speaker arrangement 20 includes an audio speaker 21, at least one controller 200, and one or more memories 700. The memory 700 may be included in the controller 200 or external to the controller 200. The controller 200 may include some or all of the transceiver module 22, the control module 24, the DSP module 26, and / or the D / A module 28. These modules 22, 24, 26, 28 may be implemented as software modules, semiconductor devices integrated in the controller 200, or a combination of both. They may also be implemented as independent components of the audio speaker arrangement 20. The memory 700 will typically include pre-stored acoustic transmission data 50 and amplitude compensation data 330 as well as software instructions so that the controller 700 can be configured to perform the method 100 for compensating the audio stream 33.

[0099] In one embodiment of the audio speaker arrangement 20, it comprises at least one audio speaker 21, at least one controller 200 and at least one memory 700. The at least one controller 200 may be configured to control the transceiver module 22 to receive an audio stream 33 and a playback volume 15 associated with the audio stream 33 through the external interface 30. The controller 200 is further configured to control the control module 24 to determine a playback sound pressure level 52 of the audio speaker arrangement 20 based on the playback volume 15 retrieved from the at least one memory 700 and pre-stored acoustic transmission data 50, so that the playback volume 15 is associated with the playback sound pressure level 52 of the audio speaker 21. The controller 200 is further configured to control the control module 24 to generate a compensation filter 320 based on amplitude compensation data 330 associated with the determined playback sound pressure level 52 retrieved from the at least one memory 700. The controller 200 is further configured to control the DSP module 26 to filter the audio stream 33 using the compensation filter 320 to generate a compensated audio stream 27. The controller 200 is also configured to control the D / A module 28 to provide the compensated audio stream 27 to the audio speaker 21 .

[0100] refer to Fig.15a and 15b , some additional details about the amplitude compensation data 330 and the hearing profile 340 will be presented. These details are specific embodiments of the amplitude compensation data 330 and the hearing profile 340, and although applicable to all related embodiments presented above, those skilled in the art will understand that these details do not limit the scope of the present disclosure.

[0101] exist Fig.15a , a graph of amplitude compensation data 330 of different frequencies is presented. The amplitude compensation data 330 is presented as a continuous line, but is preferably described as discrete amplitude compensation values ​​335 at a plurality of different frequencies. 60 degrees, Fig.15a The solid line in the example is the baseline. This means that, as shown in the reference Figure 8b As explained, a playback sound pressure level 52 corresponding to 60 phons, ie, 60 dB SPL at 1 kHz, will experience amplitude compensation data 330 of 0 dB at all frequencies. Fig.15a The 20 fang line, 40 fang line, 80 fang line, and 100 fang line are shown as the difference between the corresponding fang line and the 60 fang line. In this example, the fang is measured at 1 kHz, and therefore, for all fangs at this frequency, the amplitude compensation data 330 is 0 dB. Assume that the audio stream 33 is enjoyed at a playback sound pressure level 52 corresponding to 60 fang, and the user 40 decides to increase the playback volume 15 to a level corresponding to a 20 dB increase in the playback sound pressure level 52. The gain of the audio stream 33 will increase by 20 dB, but in addition, the 80 fang line ( Fig.15aThe frequency-dependent amplitude compensation data 330 (short dashed line in FIG. 3 ) will be added to the 20 dB gain. Therefore, the gain will be about 30 dB at low frequencies, 20 dB at 1 kHz, 19 dB at 4 kHz, and 25 dB at high frequencies.

[0102] As introduced previously, in addition to volume-related compensation, a hearing profile 340 may also be applied to the audio stream 33. An example of a hearing profile 340 is depicted in Fig.15b , where the hearing thresholds for the right ear (as represented by the crosses) and the left ear (as represented by the squares) are shown separately for a number of different frequencies. Adding hearing compensation to the compensation associated with a 20 dB change in playback sound pressure level 52 as described above, for the left ear, the gain would be about 35 dB at low frequencies, 25 dB at 1 kHz, 34 dB at 4 kHz, and 45 dB at high frequencies. The corresponding figures for the right ear would be about 40 dB at low frequencies, 35 dB at 1 kHz, 39 dB at 4 kHz, and 40 dB at high frequencies.

[0103] In the present disclosure, the inventors have solved several problems, the mapping from playback volume 15 to playback sound pressure level 52, the change of playback sound pressure level 52 and the compensation change are almost instantaneous.

Claims

1. A method (100) for real-time processing of an audio stream (33) in an audio speaker arrangement (20) comprising at least one audio speaker (21), the method (100) comprising: receiving (110) an audio stream (33) and a playback volume (15) associated with the audio stream (33), determining (120) a playback sound pressure level (52) of the audio speaker (21) based on the playback volume (15) and pre-stored acoustic transmission data (50), wherein the pre-stored acoustic transmission data (50) associates the playback volume (15) with the playback sound pressure level (52) of the audio speaker (21), generating (130) a compensation filter (320) based on amplitude compensation data (330) associated with the determined playback sound pressure level (52), and filtering (140) the audio stream (33) using the compensation filter (320) and applying a gain factor to the audio stream (33) to generate a compensated audio stream (27), wherein the gain factor is configured such that the total audio power of the audio stream (33) is maintained regardless of the compensation filter (320, 320'), wherein the gain factor is generated by estimating the signal frequency content of an average audio stream at a master level and estimating an average audio power of the average audio stream, and The compensated audio stream (27) is provided (150) to the audio speaker (21).

2. The method (100) of claim 1, wherein the amplitude compensation data (330) is based on equal loudness contours (300).

3. The method (100) of claim 1, wherein the amplitude compensation data (330) comprises amplitude compensation values ​​(335) for a plurality of different frequencies and a plurality of different playback volumes (15).

4. The method (100) according to claim 1, further comprising the following steps: receiving (110) an updated playback volume (15') different from the playback volume (15), determining (120') an updated playback sound pressure level (52') of the audio speaker (21) based on the updated playback volume (15') and the pre-stored acoustic transmission data (50), generating (130') an updated compensation filter (320) based on amplitude compensation data (330) associated with the determined updated determined playback sound pressure level (52'), filtering (140) the audio stream (33) to generate the compensated audio stream (27) using the updated compensation filter (320), and The compensated audio stream (27) is provided (150) to the audio speaker (21) using the updated compensation filter (320) in synchronization with the updated playback volume (15').

5. The method (100) according to claim 4, wherein the audio stream (33) is filtered in audio frames (23).

6. The method (100) of claim 5, wherein the steps of determining (120') an updated playback sound pressure level (52) and generating (130') an updated compensation filter (320) are performed during an intermediate time (TI) between filtering of two consecutive audio frames (23).

7. The method (100) of claim 1, wherein the step of generating (130, 130') a compensation filter (320) is further based on a hearing profile (340) associated with a user (40) of the audio speaker arrangement (20).

8. The method (100) of claim 1, wherein the audio stream (33) is formed in a channelized audio format.

9. The method (100) of claim 8, wherein the channelized audio format comprises a left channel audio stream (33L) and a right channel audio stream (33R), and wherein the step of generating (130, 130') the compensation filter (320) further comprises generating (130, 130') a left compensation filter (320) for the left channel audio stream (33L) and a right compensation filter (320) for the right channel audio stream (33R).

10. The method (100) of claim 1, wherein the compensation filter (320) is a digital filter having a linear phase response.

11. The method (100) of claim 1, wherein the compensation filter (320) is a FIR filter.

12. An audio speaker arrangement (20), comprising: at least one audio speaker (21); at least one memory (700); at least one external interface (30); and at least one controller (200) operatively connected to the at least one audio speaker (21), the at least one memory (700) and the at least one external interface (30); The at least one controller (200) is configured to perform the method (100) for real-time processing of an audio stream (33) according to any one of claims 1 to 11.

13. An audio speaker arrangement (20) according to claim 12, wherein the external interface (30) is a wireless interface.

14. A computer program product (600) configured to, when executed by a controller (200), cause the controller (200) to perform the method (100) for real-time processing of an audio stream (33) according to any one of claims 1 to 11.

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