![]() METHOD AND APPARATUS FOR PLAYING STANDARD MEDIA AUDIO AND NO INTEGRATED NOISE METADATA IN NEW MEDIA
专利摘要:
METHOD AND APPARATUS FOR PLAYING STANDARD MEDIA AUDIO WITH AND WITHOUT INTEGRATED NOISE METADATA IN NEW MEDIA DEVICESA decoder device is provided to decode the bit stream to produce an audio output signal from it, the bit stream comprises audio data and, optionally, noise metadata, containing a valuereference noise, the decoder device comprises: an audio decoder device configured to reconstruct an audio signal from audio data; and a signal processor configured to produce the audio output signal based on the audio signal; characterized in that the signal processor comprises a gain control device configured to adjust a noise level of the audio output signal; wherein the gain control device comprises a reference noise decoder configured to create a noise value, where the noise value is the reference noise value in case the reference noise value is present from the bit stream. 公开号:BR112015017295A2 申请号:R112015017295-4 申请日:2014-01-27 公开日:2020-10-20 发明作者:Robert Bleidt 申请人:Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V.; IPC主号:
专利说明:
[0001] [0001] The present invention relates to the noise control of audio, video and multimedia content digitally played on electronic reproduction devices, specifically, but not exclusively, to the reproduction noise control with content that is prepared with and without built-in noise metadata, as is commonly the case with new media devices. [0002] [0002] In the production and transmission of music, video and other multimedia content, the noise normalization process is carried out to ensure that the consumer hears the audio signal with an appropriate noise from music to music or from program to program. Since the beginning of recording and films, this has been done during the production process or through reproduction standards for cinemas. The common practice today in the music and broadcasting industries is to adjust the noise to a value close to the maximum peak level of the medium, while the practice in the film or television industries is to use one of the various noise level patterns that can be used. 20 to 3ldB below the peak peak. In the era prior to media coverage, this was not noticed by consumers, as separate devices or volume settings were used to reproduce each type of content. [0003] [0003] With the advent of mobile devices, such as cell phones or portable media players, which are designed to play music and film content, this difference in production practices leads to differences in noise that can reach 30dB, if the content is transmitted to the device without modification. This can lead to movies that are too quiet or songs that are too loud when switching from one type of content to another. [0004] [0004] A related trend is the increase in noise from many genres of recorded music through the use of compression, limitation and saturation of the dynamic range during the mastering of a recording. This mastering is done considering only lossless recording media, such as Compact Discs, although most of the music sold today is in lossy compressed data formats, like MPEG AAC and MP3. The compressed data process can introduce changes in the waveform of the reconstructed temporal domain in the decoder during playback, which causes the waveform to over-indicate the limits on a large scale or the maximum peak value of the signal. In a fixed-point decoder (or saturated floating-point decoder), typically used in mobile devices, it can lead to saturation of the over-indication to the limit on a large scale, causing additional saturation audible in the reproduced signal. [0005] [0005] The strong compression and saturation of the music are done in some cases for artistic reasons, but are more commonly done as an attempt to increase the commercial appeal of a recording by making it "sound louder than others, or to provide content that can be understood in all listening circumstances, such as airports or noisy locations, as well as in quiet environments. [0006] [0006] In the film and video industries, the wide dynamic range of audio is used in some genres for dramatic effects and to create more immersive experiences. When transmitted to the consumer via Dolby Digital or MPEG-4 AAC codecs, dynamic audio range control metadata is often included to allow the dynamic range to be optionally reduced on the receiver or device in cases where there is a noisy environment or in that noisy scenes will be very disturbing. [0007] [0007] Traditional metadata included in DVD or Blu-Ray content encoded with Dolby Digital, or transmitted by TV signals encoded with Dolby Digital (standardized in the A / 52 Audio Compression Standard of the Advanced Television Systems Committee, Inc. - Advanced Television Systems Committee) or MPEG-4 AAC (standardized in ISO / IEC 14496-3 and ETSI TC 101 154) includes the following components: [0009] [0009] Unfortunately, dynamic range control grids, as installed in lossless codecs, such as MPEG AAC or the Dolby Digital family, cannot compress a signal strongly enough to match the noise of contemporary music, depending on the metadata. they affect the common signal power (potentially in different frequency bands) on an audio compression frame basis, with common frame periods of 20-40ms. This frame-by-frame gain control is not fast enough to reduce the usual peak signal rate for this highly contemporary processed music. [0012] [0012] Therefore, it is an objective of the invention to provide a unified approach to the problem of normalization of reproductive noise and possible integrated noise metadata and podcast / radio or music content, with potentially an extremely narrow dynamic range and strong compression, limitation and saturation, but probably not containing the integrated noise metadata, due to the vast amount of previous music content already maintained or exchanged by consumers. [0014] [0014] Another objective of this invention is to avoid a potential saturation in data compression without loss in audio decoders, such as AAC, MP3 or in the Dolby Digital decoder, caused by the changes in signal components introduced by the data compression process. [0015] [0015] Another objective of this invention is to provide a moderate incentive for the music recording industry to abandon the search for compression, limitation and saturation of dynamic tracks that are always stronger in their content. [0016] [0016] However, another objective of this invention is to limit the additional workload of the device's CPU or DSP, caused by noise processing or saturation prevention. [0017] [0017] An application of the invention includes a decoder device for decoding a bit stream, so that an audio output signal, bit stream compression audio data and, optionally, noise metadata are produced from it. containing a reference noise value, the decoder device comprises: an audio decoder device configured to reconstruct an audio signal from the audio data; and an audio processor configured to produce an audio output signal based on the audio signal; characterized in that the signal processor comprises a gain of the control device configured to adjust the level of the audio output signal; wherein the gain control device comprises a noise decoder configured to create a noise value, where the noise value is the reference of the noise value in case the noise value is present in the bit stream; wherein the gain control device comprises a gain calculator configured to calculate the gain value based on a volume control value, which is provided by a user interface, allowing the user to control the volume control value; wherein the gain control device comprises a noise processor configured to control the noise of the audio output signal based on the gain value. [0019] [0019] The reference noise decoder is configured to decode the noise data contained in the bit stream. If the noise data contains the reference noise value, the reference noise decoder provides only this reference noise value as the noise value. [0023] [0023] In a preferred application of the invention, the noise value is a pre-adjusted noise value, in case the reference noise value is not present in the bit stream. These features allow for high quality bit stream reproduction without noise metadata. [0024] [0024] In a preferred application of the invention, the preset noise value is adjusted to a value between -4dB and -10dB, in particular, between -6dB and -8dB, with reference to a large scale amplitude. Empirical studies of contemporary music show that the upper noise limit for observed music content that is intended for large-scale reproduction is around -7dB. Consequently, the preset noise values, as stated, provide an optimized mode for bit stream reproduction containing no noise metadata. [0026] [0026] The dynamic range control device can compress a dynamic range calculator, which is capable of calculating a dynamic range value based on the dynamic range control value removed by the dynamic range control switch and based on a compression control value, which is provided by a user interface, allowing a user to control the compression control value. The dynamic range calculator can, in particular, be a multiplier. [0027] [0027] In addition, a dynamic range processor is provided, which is capable of controlling the dynamic range of the audio output signal based on the dynamic range value. [0029] [0029] The limiting device provides limitation with the purpose of preventing saturation from over-indication of the decoder, the volume limitation to prevent hearing loss or user preference, and artistic compression to allow the reversible generation of content with peak limitation when necessary, due to the listening environment or the user’s taste. [0030] [0030] According to the preferred application of the invention, the control component is configured to control the limiting component depending on a bit rate of the bit stream. The probability of saturation of the decoder over-indication increases when the bit rate is lowered. Therefore, the prevention of decoder over-indication saturation is enhanced when the limiting component is controlled, depending on the bit rate of the bit stream. [0031] [0031] According to the preferred application of the invention, the control component is configured to control the limiting component depending on the compression effectiveness of the audio decoder device. The effectiveness of compression of an audio encoding device that produces the bit stream and, at the same time, of the audio decoding device decoding the bit stream, describes the amount of data that is reduced when encoding the original audio data to produce the bit stream. [0034] [0034] According to the preferred application of the invention, the control component is configured to control the limiting component depending on the volume limit value set by the user or manufacturer to prevent damage to hearing. With these characteristics, hearing damage can be effectively prevented. [0038] [0038] An application of the invention includes a system comprising a decoder and an encoder, characterized in that the decoder is designed as stated. [0040] [0040] An application of the invention includes a computer program for execution, when executed on a computer or a processor, the method as declared therein. [0041] [0041] The preferred applications of the invention are subsequently discussed with respect to the attached drawings, in which: [0042] [0042] Figure 1 illustrates a block diagram of a prior art compressed data audio decoder, with support for noise metadata, as specified by ISO / IEC 14496-3 and ETS ITS 101 154, as integrated in a typical mobile phone, tablet computer, or portable media player; [0043] [0043] Figure 2 illustrates an application of a decoder with a compressed data audio decoder device and an optional audio limiter, according to the invention, which is suitable for integration into a typical mobile phone, tablet computer or portable media player; [0044] [0044] Figure 3 illustrates a function empirically provided for possible additional saturation due to the over-indication of the reconstructed signal waveform in a stereo AAC-LC decoder against the bit rate of the bit stream; [0045] [0045] Figure 4 shows a block diagram of a preferred application of the optional limiting device according to the invention; and [0046] [0046] Figure 5 illustrates a block diagram of a preferred application of the optional limiting device operating in artistic limiting mode, according to the invention. [0047] [0047] As an aid to understanding the operation of the invention, the operation of the compressed data decoding device with metadata enabled from a prior art 21, as specified by [0048] [0048] The dynamic range control switch 12 allows the application of either the light dynamic range control values, as typically used in "Linear Mode", or heavy dynamic range control values 7 'typically used in "RF Mode" , or none of them. These values 6 and 7 are sent for each frame of the compressed data bit stream for a plurality of frequency bands or regions in the bi ts stream 1 and applied to a dynamic range processor 13, which is designed as a multiplier 13, for change the output level of the audio decoder device 9, so that the short-term noise (in the order of seconds) of the audio output signal 18 is compressed according to the desired dynamic range. [0049] [0049] Metadata 3 also contains reduced gain values 5, which are used to adjust the channel mix of multichannel content (such as the 5.1 channel surround program) on a mono or stereo output when needed. [0050] [0050] Substantially, if there is no reference to the noise value 4 present in a given bit stream, the noise value 31 removed by the reference noise decoder 10 will also be adjusted to the target level of the decoder removed by the static target level provider. [0051] [0051] The output of the audio decoder 21 is then typically supplied to an audio mixing system 23, in which the audio output signal8 is combined with the user interface sounds (UI sounds), notification sounds or other audio signals 22, so that the embedded audio signal 19 is created. The overall volume is controlled by the volume control value 20. The audio signal mixing operation 23 may include secondary volume controls for adjusting the relative levels of each type of audio signal or changing its amplitude, depending on the mode of operation of the device, which is not pertinent to the operation understanding of the invention. The important thing is that the audio output signal 18 of the decoder device 21 is typically sized so that a large scale output signal corresponds to a maximum fixed point value or a large scale floating value (typically in the range -1.0 to 1.0) nominal. With strongly compressed audio data, as is typical in contemporary music, the output signal from decoder 18 will have peaks that will approximate its large scale values when heard at nominal listening levels. Thus, a large-scale peak of FS OdB (referring to the large-scale amplitude of the audio output signal) in the audio output signal 18 will be attenuated in the audio mixing system 23 and correspond to a sound pressure level (SPL ) in the listener's ear of perhaps 75dB SPL when heard in a quiet environment. [0052] [0052] Figure 2 describes a device 41 for decoding a bit stream 1, so as to produce an audio output signal 42 therefrom, bit stream 1 comprises audio data 2 and, optionally, noise metadata 3, containing a reference noise value 4, the decoder device 4 comprises: an audio decoder device 9 configured to reconstruct an audio signal 8 from the audio data 2; and a signal processor 27 configured to produce the audio output signal 42 based on the audio signal 8; characterized in that the signal processor 27 comprises a gain control device 10, 15, 28 configured to adjust an audio output signal level 42; wherein the gain control device 10, 15, 28 comprises a reference noise decoder 10 configured to create a noise value 37, where the noise value 37 is the reference noise value 4, in the case that the reference noise value 4 is present in bit stream 1; wherein the gain control device 10, 15, 28 comprises a gain calculator 28 configured to calculate the gain value 33 based on the noise value 37 and based on the volume control value 20, which is provided by a user interface, allowing the user to control the volume control value 20; wherein the gain control device 10, 15, 28 comprises a noise processor 28 configured to control the noise of the audio output signal 42 based on the gain value 33. [0053] [0053] The audio decoding device 9 can be any device 9, which is able to reconstruct the audio signal 8 from the audio data 2 of the compressed bit stream. The signal processor 37 can be any device 37, which is capable of producing the audio output signal 42, when the audio signal 8 of the audio decoder device 9 is fed to it, and which has a gain control device 10, 15, 28, as explained below. The gain control device 10, 15, 28 is a device that is adjusted to control the noise of the audio output signal 42. [0054] [0054] The noise decoder 10 is configured to decode the noise data 3 contained in the bit stream 1. If the noise data 3 contain a reference noise value 4, the reference noise decoder removes only this value from reference noise 4 as a noise value 37. [0058] [0058] In a preferred application of the invention, the noise value 37 is a pre-adjusted noise value 37, in case the reference noise value is not present in the flow of bi ts 1. These characteristics allow a high reproduction quality of bit stream 1 without noise metadata 3. [0059] [0059] In a preferred application of the invention, the preset noise value 37 is adjusted to a value between - 4dB and -10dB, in particular, between -6dB and -8dB, with reference to a large scale amplitude. Empirical studies of contemporary music show that the upper noise limit for observed music content, which is intended for large-scale reproduction, is around -7dB. [0060] [0060] In a preferred application of the invention, signal processor 27 comprises a dynamic range control device 12, 13, 14 configured to adjust the dynamic range of the audio output signal 42, characterized by the dynamic range control device 12, 13, 14 comprise a dynamic range control switch 12 configured to receive at least one dynamic range control value 6, 7 from noise metadata 3 and to alternatively withdraw one of the dynamic range control values 6, 7 received or a preset dynamic range control value 43, wherein the dynamic range control device 12, 13, 14 comprises a dynamic range calculator 14 configured to calculate a dynamic range value 44 based on the dynamic range control value 6, 7, 43 taken by dynamic range control switch 12 and based on a compression control value 25, which is provided by a user interface, allowing the user ary control the compression control value 25; wherein the dynamic range control device 12, 13, 14 comprises a dynamic range processor 13 configured to control the dynamic range of the audio output signal 42 based on the dynamic range value 44. [0061] [0061] Dynamic range control device 12, 13, 14 comprises a dynamic range control switch 12, which is configured to the noise metadata decoder 3 of the bit stream lt in such a way that, at least , a dynamic range control value 6, 7 can be provided. [0062] [0062] The dynamic range control device 12,13, 14 can comprise a dynamic range calculator 14, which is capable of calculating a dynamic range value 44 based on the dynamic range control value 6, 7, 43 taken by dynamic range control switch 12 and based on a compression control value 25, which is provided by a user interface, allowing a user to control the compression control value 25. The dynamic range calculator 14 can, in especially, be a multiplier 14. [0063] [0063] In addition, a dynamic range processor 13 is provided, which is capable of controlling the dynamic range of the audio output signal 42 based on the dynamic range value 44. With these characteristics, the reproduction of the bi stream ts 1 can be adapted to the listening environment and / or the taste of the listener. [0064] [0064] Figure 2 illustrates the operation of a preferred application of the invention, as contained in an improved audio decoder 41. The audio input bit stream 1 consists of essential audio data 2 and optional noise metadata 3 containing the above mentioned standard metadata values for the program reference level 4, reduction gains 5, light CKD values 6 and heavy CKD values 7. Metadata 3 can also include artistic limiting parameters 32 and true peak values 36, which are used in an optional application. [0069] [0069] Those skilled in the art will understand that it may be necessary to balance or increase the volume control value 20, an audio mixing system 23 or a subtractor 28, so that the volume of the mixed audio signal 29 is located in the noise , with the supplementary audio noise signal set 24. [0070] [0070] In previous approaches, to match the [0071] [0071] In contrast, the present invention does not require a limiter to control the peak for a normal range of 42t audio output signal for the purpose of noise matching, but may include an optional limiting device for the purpose of saturation protection, for limitation to avoid hearing damage, and for limitation of artistic effect or increased compression. A special decoder device 41 can be equipped with the limiting device 30 for any or all of these purposes, with varying application costs, or the decoder device 30 can simply be omitted. Each of these cases is explained below. [0072] [0072] When considering the case of protection against saturation, two sub-cases of signals must be considered: Some data streams l may not contain metadata 3, such as legacy music content already presented on the user's device, which was not analyzed for noise or dynamic range. In this sub-case, multiplier 13 is not active and multiplier 15 provides a maximum unit gain in the maximum volume control configuration. Thus, the only potential for saturation is the possibility of induced compression over-indication in the signal waveform. The amount of potential over-indication possible with normal signals can be empirically determined by a compression codec at a confidence interval, while a function of bits per second, per channel or similar metric compression range. A typical empirically determined saturation prediction function 56 for AAC LC stereo bit stream is shown in Figure 3. It should be understood by those skilled in the art, that these other empirical, analytical or interactive methods can be used to determine or predict the quantity of saturation that would be present. [0074] [0074] The limiting device 30 provides limitation with the objective of preventing saturation from over-indication of decoding, volume limitation to prevent loss of listening or user preference, and artistic compression, to allow the reversible generation of content with peak limitation , when necessary, due to the listening environment or taste of the user. [0075] [0075] The limiting device 51 is controlled by internal signals or provided with peak or artistic level metadata, which provides limitation for the purpose of preventing over-indication of the decoder, volume limitation for preventing hearing loss or user preference, and compression artistic, to allow the reversible generation of content with peak limitation, when necessary, due to the listening environment or taste of the user. [0076] [0076] Limiter 51 is ideally an efficient, non-saturation, preventive limiter as it is commonly used for digital audio mastering and known by experts in the art. For example, it can be an application as described in [8]. Alternatively, if saturation protection is not a desired feature, and the volume limitation is a hard saturation with threshold adjustment by output 58, it can be replaced and the buffer 53 is removed or shortened. [0079] [0079] According to the preferred application of the invention, the control component is configured to control the limiting component 62, depending on the compression effectiveness of the audio decoding device 9. The compression effectiveness of an audio encoding device producing the flow bit and at the same time the audio decoder device 9 decoding the bit stream 1, describes how much the amount of data is reduced by encoding the original audio data, to produce bit stream 1. The greater the amount of data reduced, the probability of saturation of over-indication of the decoder increases. Therefore, the saturation prevention of the decoder over-indication is increased when the limiting component 62 is controlled, depending on the efficiency of the audio decoder device 9. [0080] [0080] In a preferred application of this optional feature, the compression effectiveness of the audio decoder device 9 is inserted into a saturation prediction device 54, which comprises a saturation prediction function 56 applied to logical confirmations or ports, for a search table, or by other techniques of applying the function of at least one variable as will be known by experts in the art. The output of function 56 is fed through function 59 minimum, [0081] [0081] In cases where the maximum output signal level of the processed core decoder 35 is less than the level predicted by the saturation prediction function 56, there is no possibility of saturation due to the over-indication of the decoder (within the range relay or fault link of function 54) and switch 52 selects the output for compensation buffer 53. The aforementioned buffer is merely a delay to match the delay processing of limiter 51, and will introduce only one workload insignificant computational, compared to the significant workload of the limiter 51. [0082] [0082] According to a preferred application of the invention, the control component 63 is configured to control the limiting component 62, depending on the gain value 33 of the gain control device 10, 15, 28. The maximum possible peak level of the audio output signal 42 is determined in this sub case by the gain value 33 of the gain control device 10, 15, 28. If the above value is OdB, the decoder device 41 is operating at its large scale limits, as commanded maximum setting of the volume control value 20. [0083] [0083] In this sub-case, where no metadata 3 is present, switch 60 removes an OdBFS value as if it were as much as possible in the audio data inputs 2 of the bit stream. [0084] [0084] According to the preferred application of the invention, the control component 63 is configured to control the limiting component 62, depending on a true peak value 36 transmitted in the noise metadata 3 of bit stream 1, and indicating a level peak power of an audio source converted to bit stream 1 by an external encoder. The use of this true peak value 36 allows the computation of a more accurate value for the maximum possible peak level of the audio output signal 42. [0085] [0085] In this case, where the data streams contain noise metadata 3, metadata 3 can be specified to also include the true peak measurement specified by the ITU BS.1770-3 standard. In this sub-case, switch 60 selects the true peak value 36 contained in noise metadata 3, instead of the constant OdB FS. The total gain adjustment 33 and the true peak value 36, indicating the maximum peak amplitude of the signal input 35 for limiter 30, is computed by the adder 61 and is then compared to the output of the saturation function 56 by the comparator [0086] [0086] According to the preferred application of the invention, the control component 63 is configured to control the limiting component 62, depending on the volume limit value 57, adjusted by the user or the manufacturer to avoid hearing damage. With these characteristics, damage to hearing can be prevented efficiently. [0087] [0087] In the case of limitation to avoid hearing damage, the user or manufacturer of the device should adjust the maximum peak level 57 to which the output should be limited using a volume limit signal. When switch 58 is activated to activate this volume limiting feature, minimum function 59 selects the lowest of the output levels required to engage limiter 51 or to limit the output, due to saturation prevention or volume limiting. The output of switch 58 is also inserted in limiter 51 to adjust its threshold to the appropriate level. [0088] [0088] According to the preferred application of the invention, illustrated in Figure 5, the control component 63 is configured to control the limiting component 62, depending on artistic limitation parameters 32, transmitted in the noise metadata 3 of the bit stream 1 and indicating the artistic limitation threshold values 74, the artistic limitation attack time values 7 4b and / or the artistic limitation time release values 7 4c. [0089] [0089] In this mode, the noise metadata 3 includes the artistic limitation parameters 32, illustrated in the bus electrical note in Figure 5, which are sent to each audio frame of the content. Contained in 32 are the limiter attack time values, the release and threshold time for the light and heavy modes selected by switch 12 and selected by a corresponding combined switch 73 to remove bus 74. Bus 74 contains the value selected artistic limitation threshold 74a, which is added to the gain setting of decoder 33 by adder 71, and the desired attack and release times 7 4b and 7 4c, which are provided directly to limiter 51. The minimum function 7 2 is used to select the Volume Limit 57 (OdB FS, if no volume limit was used) or the output of the adder 71. In this way, the limiter 51 normally operates at a threshold controlled by the value 74a until the volume control 20 is increased to a point where the volume limit is reached and limit the maximum level of the limiter threshold. In this mode, limiter 51 operates continuously and switch 52 is always in the display position. The artistic use of these parameters can be achieved by monitoring the output of a device, audio software plug-in, or other device that contains a copy of the invention, during mixing, mastering or other creative operations or distributions. [0090] [0090] According to the preferred application of the invention, there is no possibility of applying the mask again after the limiting device 30 artificially increases its noise, as this would remove the slight incentive mentioned above. [0091] [0091] According to the preferred application of the invention, the control component 63 is configured to control the limiting component 62 continuously or repeatedly. These characteristics allow variable control of the limiting component 62 over time. [0092] [0092] According to the preferred application of the invention, the limiting device 30 is configured to bypass the limiter 51 through a bypass device 53, having a transfer function, which is, regarding gain and delay, similar to the function transfer of limiter 51. With these characteristics, the workload of signal processor 27 can be reduced significantly. [0093] [0093] Experts in the art will understand that this process can be applied in software such as serial computer instructions or in hardware components. The operations described here are typically performed as software instructions by a computer CPU or Digital Signal Processor, and the records and operations illustrated in the figures can be applied by corresponding computer instructions. However, this does not exclude the application in an equivalent hardware design, using hardware components. Also, it will be understood by experts in the art that the values 4, 6, 7, 2 O, 33, 3 6, 57, 74a, and others, will typically be expressed in a domain of logarithmic scale, according to the practical norm and specified in the standards referenced. Furthermore, the operation of the invention is displayed here in a sequential and elementary manner. [0097] [0097] Depending on certain application requirements, intervention applications can be applied in hardware or in software. The application can be performed using non-transitory storage media, such as digital storage media, for example, a floppy disk, a DVD, a Blu-Ray, a CD, a ROM, a PROM and EPROM, an EEPROM or a memory FLASH, having electronically readable control signals stored in them, which cooperate (or are capable of cooperating) with a programmable computer system, such that the respective method is carried out. [0098] [0098] Some applications, according to the invention, comprise a data carrier with electronically readable control signals, which are capable of cooperating with a programmable computer system, such that one of the methods described in this document is performed. [0099] [0099] Generally, the applications of the present invention can be applied with a computer program product with a programming code, the programming code is practical to perform one of the methods, when the computer program product is run on a computer . The programming code can, for example, be stored on a machine with readable support. [0101] [0101] In other words, an application of this inventive method is, therefore, a computer program having a programming code for carrying out one of the methods described in this document, when the computer program is executed on a computer. [0103] [0103] Another application of the method of invention is, therefore, a stream of data or sequence of signals representing a computer program for carrying out one of the methods described in this document. The data stream or signal sequence can, for example, be configured to be transferred via a data communication connection, for example, via the internet. [0104] [0104] Another application comprises processing means, for example, a computer or a programmable logic device, configured to, or adapted to, perform one of the methods described in this document. [0105] [0105] Another application comprises a computer having the computer program installed to perform one of the methods described in this document. [0106] [0106] Another application, according to the invention, comprises an apparatus or a system configured to transfer (for example, electronically or optically) a computer program for carrying out one of the methods described in this document. A receiver can, for example, be a computer, a mobile device, a memory device or the like. The apparatus or system can, for example, compress a file server to transfer the computer program to the receiver. [0107] [0107] In some applications, a programmable logic device (for example, an array of programmable field gates) can be used to perform some or all of the functionality of the methods described in this document. In some applications, an array of programmable field gates can cooperate with a microprocessor to perform one of the methods described in this document. Generally, the methods are preferably performed by any hardware device. [0108] [0108] The applications described above are merely illustrative for the principles of the present invention. It is understood that the modifications and variations of the arrangements and details described in this document will be apparent to those skilled in the art. It is therefore the intention to be limited only by impeding the scope of patent claims and there are specific details presented in the manner described and explained in the applications in this document. REFERENCE SIGNS: [2] [2] European Telecommunications Standards Institute, ETSI TS 101 154: Digital Video Broadcasting (DVB); Specification for the use of Video and Audio Coding in Broadcasting Applications based on the MPEG-2 transport stream, www.etsi.org, [3] [3] Advanced Television Systems Committee, Inc., Audio Compression Standard A / 52, www.atsc.org. [4] [4] International Telecommunications Union, Recommendation ITU-R BS .1770-3: Algorithms to measure audio program loudness and true-peak audio level, www.itu.int. [6] [6] Florian Camerer, et al, "Loudness Normalization: The Future of File-Based Playback," Music Loudness Alliance, www.music-loudness.com. [7] [7] Dolby Laboratories, Inc., Dolby Digital Professional Encoding Guide-lines, www.dolby.com. [8] [8] Perttu Hamalainen, "Smoothing Of The Control Signal Without Clipped Output In Digital Peak Limiters", Proc. of the 5th International Conference on Digital Audio Effects, Hamburg, Germany, September 26-28, 2002.
权利要求:
Claims (16) [1] 1. A decoder device for decoding a bit stream (1) to produce an audio output signal (42) from there, the bit stream (1) comprises audio data (2) and, optionally, metadata from noise (3), containing a reference noise value (4), the decoder device (41) comprises: an audio decoder device (9) configured to reconstruct an audio signal (8) from audio data (2); and a signal processor (27) configured to produce the audio output signal (42) based on the audio signal (8); characterized in that the signal processor (27) comprises a device for gain control (1, 15, 28) configured to adjust a noise level of the audio output signal (42); in which the fact of the gain control device (10, 15, 8) comprises a reference noise decoder (10) configured to create noise value (37), in which the noise value (37) is the reference noise value (4) if the reference noise value (4) is present from the bit stream (1); in which the fact of the gain control device (10, 15, 28) comprises a gain calculator (28) configured to calculate the gain value (33) based on the noise value (37) and based on the volume control value (20), which is provided by the user interface, allowing the user to control the volume control value (20); wherein the fact of the gain control device (10, 15, 28) comprises a noise processor (15) configured to control the noise level of the audio output signal (42) based on the gain value (33) . [2] 2. The decoding device, according to the previous claim, characterized by the fact that the noise value (33) is a pre-adjusted noise value, in case the reference noise value (4) is not present in the flow of bits (1). [3] 3. The decoder device, according to the previous claim, characterized by the fact that the pre-adjusted noise value is adjusted to a value between - 4dB and -10dB, in particular, between -6dB and -8dB, referring to an amplitude large-scale. [4] 4. The decoder device according to the previous claim, characterized in that the signal processor (27) comprises a dynamic range control device (12, 13, 14) configured to adjust the dynamic range of the audio output signal (42), where the dynamic range control device (12, 13, 14) comprises a dynamic range control switch (12) configured to provide at least one dynamic range value (6, 7) the noise metadata (3) and to alternatively remove one of the supplied dynamic range control values (6, 7) or a dynamic range control value (43); wherein the fact that the dynamic range control device (12, 13, 14) comprises a dynamic range calculator (14) configured to calculate a dynamic range value (44) based on the dynamic range control value ( 6 '7' 43) removed by the dynamic range control switch (12) and based on a compression control value (25), which is provided by a user interface, allowing the user to control the control value of compression (25); wherein the dynamic range control device (12, 13, 14) comprises a dynamic range processor (13) configured to control the dynamic range of the audio output signal (42) based on the dynamic range value ( 44). [5] 5. The decoder device according to the preceding claim, characterized in that the signal processor (27) comprises a limiting device (30) configured to limit an amplitude of the audio output signal (42), in which the limiting device (30) comprises a limiter component (62) having a limiter (51) in a control component (63) configured to control the limiter component (62), wherein a processed audio signal (35), which is provided by the signal audio (8) when being processed, at least, by the gain control device (10, 15, 28), it is inserted by the limiting component (62), and in which the audio output signal (42) is removed by the limiting component (62). [6] 6. The decoder device according to the preceding claim, characterized in that the control component (63) is configured to control the limiting component (62), depending on the bit rate of the bit stream (1). [7] 7. The decoder device, according to claims 5 or 6, characterized in that the control component (63) is configured to control the limiting component (62), depending on the effectiveness of the audio decoder device (9). [8] 8. The decoder device according to one of claims 5 to 7, characterized in that the control component (63) is configured to control the limiting component (62), depending on the true peak value (36) transmitted in the noise metadata (3) of the bit stream (1) and indicate a maximum peak level of an audio source converted to the stream of bi ts (1) by an external encoder. [9] 9. The decoder device according to claims 5 to 8, characterized in that the control component (63) is configured to control the limiting component (62), depending on the gain value (33) of the control device gain control (10, 15, 28). [10] 10. The decoder device according to claims 5 to 9, characterized in that the control component (63) is configured to control the limiting component (62), depending on the volume limit value (57) set by the user or manufacturer to avoid hearing damage. [11] The decoder device according to one of claims 5 to 10, characterized in that the control component (63) is configured to control the limiting component (62) depending on the artistic limiting parameters (32) transmitted in the noise metadata ( 3) of the bit stream (1) and indicate threshold values for artistic limiters (74a), attack time values for artistic limiters (74b) and / or release time values for artistic limiters (74c). [12] The decoder device according to one of claims 5 to 11, characterized in that the control component (63) is configured to control the limiting component (62), continuously or repeatedly. [13] 13. The decoder device according to one of claims 5 to 12, characterized in that the limiting device (30) is configured to bypass the limiter (51) by means of a bypass device (53) having a function transfer, which is, in relation to gain and delay, similar to the transfer function of the limiter (51). [14] 14. A system comprising a decoder device (41) and an encoder, characterized in that the decoder device (41) is designed according to one of claims 1 to 13. [15] 15. A method of decoding bi-stream (1) to produce an audio output signal (42) from there, the bit stream (1) comprises audio data (2) and, optionally, metadata from noise (3), containing a reference noise value (4), the method comprises the steps: reconstructing the audio signal (8) from the audio data (2) using an audio decoding device (9); and producing the audio greeting signal (42) based on the audio signal (8) using a signal processor (27); characterized by the fact that a noise level of the audio output signal (42) is adjusted using a gain control device (10, 15, 28) compressed by the signal processor (27); where the fact of the noise value (37) is created by a reference noise decoder (10) compressed by the gain control device (10, 15, 28), where the noise value (37) is the value reference noise (4) if the reference noise value (4) is present in the bit stream; where the fact of a gain value (33) is calculated based on the value (3 7) and based on the volume control value (20), which is provided by a user interface, allowing the user to control the volume control value (20) by a gain calculator (28) compressed by the gain control device (10, 15, 28); wherein the fact of the noise level of the audio output signal (42) is controlled based on the gain value (33) by a noise processor (15) compressed by a gain control device (10, 15, 28 ); [16] 16. The computer program for carrying out, when run on a computer or processor, characterized by the method of claim 15.
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同族专利:
公开号 | 公开日 RU2015136531A|2017-03-07| US20150332685A1|2015-11-19| CN110853660A|2020-02-28| MX351187B|2017-10-04| TWI524330B|2016-03-01| KR20150109418A|2015-10-01| CN105190750A|2015-12-23| EP2948947A1|2015-12-02| ES2628153T3|2017-08-01| KR101849612B1|2018-04-18| CA2898567C|2018-09-18| JP6445460B2|2018-12-26| TW201438003A|2014-10-01| RU2639663C2|2017-12-21| AR096574A1|2016-01-20| CN105190750B|2019-10-25| CA2898567A1|2014-07-31| US9576585B2|2017-02-21| JP2016509693A|2016-03-31| MX2015009534A|2015-10-30| EP2948947B1|2017-03-29| WO2014114781A1|2014-07-31|
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法律状态:
2018-11-13| B06F| Objections, documents and/or translations needed after an examination request according [chapter 6.6 patent gazette]| 2020-11-10| B06U| Preliminary requirement: requests with searches performed by other patent offices: procedure suspended [chapter 6.21 patent gazette]| 2021-11-23| B350| Update of information on the portal [chapter 15.35 patent gazette]| 2021-12-07| B07A| Application suspended after technical examination (opinion) [chapter 7.1 patent gazette]|
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申请号 | 申请日 | 专利标题 US201361757606P| true| 2013-01-28|2013-01-28| US61/757,606|2013-01-28| PCT/EP2014/051484|WO2014114781A1|2013-01-28|2014-01-27|Method and apparatus for normalized audio playback of media with and without embedded loudness metadata on new media devices| 相关专利
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