KR101583293B1 - Method and apparatus for controlling audio signal loudness - Google Patents

Abstract

A method for controlling an audio signal size is disclosed. The control method includes applying a first gate block and a second gate block having a predefined gate size to an input audio signal to generate a first gate signal corresponding to a size of a first audio signal corresponding to a second gate block, Measuring a size of the second audio signal, calculating a first gate weight corresponding to the first gate block using the measured size of the first audio signal, and using the measured size of the second audio signal, Detecting a frame in which a gate handover has occurred in the first gate block and the second gate block, calculating a second gate weight corresponding to the detected frame using the calculated first and second weights, Interpolating the frame weights, and interpolating the frame weights, the first and second gate weights, Performing a mailing by a step of controlling the magnitude of the audio signal.

Description

[0001] The present invention relates to a method and apparatus for controlling audio signal size,

The present invention relates to a method and apparatus for controlling the size of an audio signal reproduced in multimedia.

People are exposed to various sounds while they are living in various environments. As shown in FIG. 1, when the human being is exposed to various kinds of sounds, various kinds of sounds are generated, such as environmental noises that cause unpleasantness when people listen, multimedia sounds and music that entertain people, And the sound that occurs when it occurs.

Different sounds around people can cause pain, joy, and a variety of information depending on the size and type of sound. This is because the human auditory structure perceives sound through the sound pressure level of the sound delivered to the air, so that the sound size and intensity are useful figures that define the auditory fatigue and physical characteristics of the sound.

Among the methods of evaluating sound, loudness is the subjective sound size perceived by the human auditory system when a sound is delivered to the human ear, and the intensity of the sound is the intensity of the objective sound transmitted to the human auditory system Which means the power of sound, usually measured in decibels well known. Typically, conversations between people are between 60 and 70 dB, traffic is heavy, noise is about 80 dB, and people generally feel comfortable in the range of about 70 dB.

Referring to FIG. 1, as the portable multimedia audio device develops, it is possible to enjoy multimedia contents and music of the present invention anytime, anywhere and in any situation. In particular, with the advent of MP3 (MPEG-1 Layer III) in the late 1990s and the popularization of the Internet in the late 1990s, it became possible to easily download and listen to MP3 compressed digital music through the Internet.

The commercial audio source market has rapidly expanded with the popularization of multimedia devices, and the audio source has become more competitive in the area, so that the dynamic range of the maximum reproducible sound and the minimum sound difference of the audio source And the maximum value of the waveform was increased, so that the audio sound size was significantly increased. This is further deepened by the idea that the larger the audio sound size, the more people will recognize it as good music.

FIG. 2 (a) shows the waveform of 1970's music (Pops), and FIG. 2 (b) shows the waveform of the Korean K-Pops in 2011. Referring to FIG. 2, it can be seen that the music recorded long ago is wider than that of the recently released sound source. In the case where the waveform of the K-pops sound source, which has recently become popular in the world, reaches a maximum value or exceeds a maximum value .

Accordingly, a technique for accurately measuring the sound size of the audio in the multimedia device, adjusting the sound size, and a technique for controlling the audio sound size are required.

The present invention provides an apparatus and a method that can control the size of an audio signal according to a standard while preventing gate delay caused by interpolation of gate weights.

According to another aspect of the present invention, there is provided a method of controlling an audio signal size, the method comprising: applying a first gate block and a second gate block having a predefined gate size to an input audio signal, Measuring a size of a second audio signal corresponding to the second gate block, the size of the first audio signal corresponding to the gate block, measuring the size of the second audio signal corresponding to the first gate block, Calculating a first gate weight, calculating a second gate weight corresponding to the second gate block using the measured magnitude of the second audio signal, calculating a second gate weight corresponding to the gate in the first gate block and the second gate block, Detecting a frame in which a handover has occurred, calculating a frame weight from the detected frame using the calculated first and second weights Teopol step of illustration and perform scaling on the input audio signal by using the interpolation of the frame weight, the first and second gate weights includes the step of controlling the size of the audio signal.

The second gate block may be a gate block shifted by overlapping a predefined size within the first gate block.

In addition, the first and second gate blocks include at least one frame, and the frame can determine a data size to be received at a time.

In addition, the number of the interpolated frame weights may be variable.

According to another aspect of the present invention, there is provided an apparatus for controlling an audio signal size according to an embodiment of the present invention. The apparatus includes a first gate block and a second gate block having a predefined gate size, A size of the first audio signal corresponding to the first gate block; an audio signal size measuring unit measuring the size of the second audio signal corresponding to the second gate block; A weight calculating unit for calculating a first gate weight corresponding to one gate block and calculating a second gate weight corresponding to the second gate block using the measured magnitude of the second audio signal, And a detector for detecting a frame in which the gate handover has occurred in the second gate block, And an audio signal size control unit for interpolating a frame weight from the detected frame and scaling the input audio signal using the interpolated frame weight and the first and second gate weights to control the size of the audio signal do.

The second gate block may be a gate block shifted by overlapping a predefined size within the first gate block.

In addition, the first and second gate blocks include at least one frame, and the frame can determine a data size to be received at a time.

In addition, the number of the interpolated frame weights may be variable.

According to various embodiments of the present invention described above, gate weights may be interpolated from a frame in which a gate handover occurs to prevent gate delay caused by interpolation of gate weights.

In addition, the number of times the gate weights are interpolated can be variably controlled.

Fig. 1 is a diagram for explaining various auditory fatigue factors occurring in daily life.
2 is a diagram showing examples of waveforms of an audio signal.
3 is a diagram for explaining a distortion phenomenon according to audio data clipping.
4 is a diagram for explaining auditory loss due to audio and noise.
5 is a diagram for explaining the normalization of the audio signal size of a digital broadcast program.
6 is a diagram illustrating a method of measuring the size of an audio signal.
7 is a graph showing an example of a frequency response characteristic of a pre-filter.
8 is a graph showing an example of the frequency response characteristic of the RLB filter.
9 is a diagram for explaining an example of a structure of a broadcast system for recording and a previously produced broadcast program.
10 is a diagram showing a first embodiment of a method of controlling the size of an audio signal.
11 is a diagram for explaining a first embodiment of a method for controlling the size of an audio signal.
12 is a diagram illustrating a basic structure of a peak value based loudness control ratio calculation for adjusting an audio signal size.
13 is a diagram showing an example of a structure of a real-time broadcasting system.
14 is a diagram showing a second embodiment of a method for controlling the size of an audio signal.
FIG. 15 is a diagram for specifically explaining a second embodiment of a method for controlling the size of an audio signal.
16 is a diagram for explaining a method in which a live LD control step is added to the final stage of the first embodiment and the second embodiment.
17 is a diagram showing a third embodiment of a method for compensating sound quality degradation due to size control of an audio signal.
18 is a diagram showing a fourth embodiment of a method for controlling the size of an audio signal in a terminal.
FIG. 19 is a flowchart specifically illustrating a method of controlling the audio signal size of the audio signal size control apparatus according to the first embodiment of the present invention.
20 is a view for explaining an audio signal size measurement method to which the audio gating method described in ITU-R 1770-2 is added.
21 is a view for explaining a gate handover in order to explain an audio signal size control method according to a fifth embodiment of the present invention.
22 is a view for explaining an audio signal size control method according to the fifth embodiment of the present invention.
23 is a view for explaining linear interpolation, which is an example of interpolation according to the fifth embodiment of the present invention.
24 to 26 are diagrams comparing the waveforms of the input audio signal and the normalized audio signal.

The following merely illustrates the principles of the invention. Thus, those skilled in the art will be able to devise various apparatuses which, although not explicitly described or shown herein, embody the principles of the invention and are included in the concept and scope of the invention. Furthermore, all of the conditional terms and embodiments listed herein are, in principle, only intended for the purpose of enabling understanding of the concepts of the present invention, and are not to be construed as being limited to such specifically recited embodiments and conditions do.

It is also to be understood that the detailed description, as well as the principles, aspects and embodiments of the invention, as well as specific embodiments thereof, are intended to cover structural and functional equivalents thereof. It is also to be understood that such equivalents include all elements contemplated to perform the same function irrespective of the currently known equivalents as well as the equivalents to be developed in the future, i.e., the structure.

Thus, for example, it should be understood that the block diagrams herein represent conceptual views of exemplary circuits embodying the principles of the invention. Similarly, all flowcharts, state transition diagrams, pseudo code, and the like are representative of various processes that may be substantially represented on a computer-readable medium and executed by a computer or processor, whether or not the computer or processor is explicitly shown .

The functions of the various elements shown in the figures, including the functional blocks depicted in the processor or similar concept, may be provided by use of dedicated hardware as well as hardware capable of executing software in connection with appropriate software. When provided by a processor, the functions may be provided by a single dedicated processor, a single shared processor, or a plurality of individual processors, some of which may be shared.

Also, the explicit use of terms such as processor, control, or similar concepts should not be interpreted exclusively as hardware capable of running software, and may be used without limitation as a digital signal processor (DSP) (ROM), random access memory (RAM), and non-volatile memory. Other hardware may also be included.

In the claims hereof, the elements represented as means for performing the functions described in the detailed description include all types of software including, for example, a combination of circuit elements performing the function or firmware / microcode etc. , And is coupled with appropriate circuitry to execute the software to perform the function. It is to be understood that the invention defined by the appended claims is not to be construed as encompassing any means capable of providing such functionality, as the functions provided by the various listed means are combined and combined with the manner in which the claims require .

BRIEF DESCRIPTION OF THE DRAWINGS The above and other objects, features and advantages of the present invention will become more apparent from the following detailed description of the present invention when taken in conjunction with the accompanying drawings, in which: There will be. In the following description, well-known functions or constructions are not described in detail since they would obscure the invention in unnecessary detail.

Hereinafter, a preferred embodiment of the present invention will be described in detail with reference to the accompanying drawings.

3 is a diagram for explaining a distortion phenomenon according to audio data clipping.

If the waveform of the sound source exceeds the allowable range of data resolution in the digital data, the waveform of the sound source is reduced, and this phenomenon is audio data clipping.

3 (a) shows the sine wave without clipping, (b) shows the waveform frequency characteristics without clipping, (c) shows the sine wave with clipping, and (d) shows the frequency characteristics of the clipping waveform.

Referring to FIG. 3, when the audio data clipping phenomenon is caused to distort the audio signal and the frequency characteristics of a simple sine waveform (FIG. 3 (B)) and the clipped sine waveform (FIG. 3 , It can be seen that a signal distortion component which was not present in the non-clipping sinusoidal waveform as shown by the dotted line in FIG. 3 (d) is generated by audio data clipping.

On the other hand, the problem caused by the increase in the audio sound size is being amplified by popularization of portable multimedia devices. Currently, adolescents with significantly increased audio listening time by multimedia devices are continuously exposed to sound sources with a fairly large audio sound size for a long time.

Referring to FIG. 4, it can be seen that the auditory loss of American teenagers significantly increased when portable multimedia devices were popularized in the mid-2000s compared with before the introduction of MP3-based portable multimedia devices in the early 1990s.

In addition, the number of patients with noise-induced hearing loss increased by about 50% in the early and late 2000s, and auditory fatigue due to multimedia devices and noise environment exceeded the threshold value, affecting auditory function deterioration.

Therefore, it is necessary to reduce the auditory fatigue caused by audio in order for people to live and enjoy safe and pleasant audio and music life.

For that purpose, one embodiment of the present invention relates to a method for accurately measuring the audio sound size and adjusting the sound size in a multimedia device.

5 is a diagram for explaining the normalization of the audio signal size of a digital broadcast program.

In Korea, efforts are being made to reduce the difference in audio signal loudness between broadcasting stations and contents through revision of the broadcasting law. The program transmitted from the current broadcasting shows a considerable size difference between broadcasting companies and broadcasting contents.

Referring to FIG. 5, the audio signal sizes (for example, Channel 1: -23.4 LKFS, Channel 2: -8.5 LKFS) of the two music contents are significantly different. These differences are causing considerable inconvenience to broadcast viewers. In order to overcome this, standardization work is proceeding under the title of "Digital broadcasting program volume level standard" under TTA PG803 WG8034.

The goal of the standardization is to adjust the channel / broadcast program having a considerable size difference according to the standardized volume standard to have a normalized audio signal size (for example, Channel 1: -24 LKFS, Channel 2: -24 LKFS) It is to prepare the output standard.

Since the standardization will be linked with the broadcasting law, if the importance and usability of the standard is considerably high, the standard will be based on the international audio signal size measurement standard ITU-1770-1 / 2, And will analyze the current digital broadcast signal size and technologies that can help to comply with it.

6 is a diagram illustrating a method of measuring the size of an audio signal.

 A study on the method of measuring the size of audio signal was started in the mid 2000s and ITU-R BS, which is the standard for measurement of audio signal size in ITU. ITU-R BS 1770-1 was announced in 2006 and added Gating method. The 1770-2 was announced in 2011.

In the published standard, only the method of measuring the audio signal size and the method of measuring the true peak are presented, and the portion of the audio signal size control is not done. So far, no standardization has been done on how to control the audio signal size.

The standardized audio signal size measurement method in ITU-R is measured through LKFS (loudness, K weighted, relative to nominal full scale) as shown in FIG.

 The first module of the algorithm (Pre-filter) is configured as a second-order IIR filter to take into account the acoustic effects of the human head.

7 is a graph showing an example of a frequency response characteristic of a pre-filter.

As shown in FIG. 7, the frequency characteristics of the filter are such that the region of 1 kHz or less is removed based on about 1 kHz, and the frequency is passed through the region of 1 kHz or more. The filter coefficients for the most commonly used 48 kHz data are based on the spherical head model of the ITU-R BS. 1770-1.

8 is a graph showing an example of the frequency response characteristic of the RLB filter.

In the second module (RLB filter), a weight filter based on human auditory characteristics is applied. This filter is based on the characteristic that the human auditory sense has a different sensitivity in the frequency domain with respect to the inputted sound as shown in Fig. 8 (a).

For example, in FIG. 8 (a), about 10 dB at 250 Hz and about 1 dB at 1 kHz based on the minimum level are perceived by a person to be the same audio sound size. Thus, a band-specific weighted filter is designed so that the filter response to account for human hearing has a filter response similar to the inverse of the same audio sound size contour defined in ISO 226, as shown in Figure 8 (b).

The weights of the low - frequency domain are designed to be lower than those of the low - frequency domain. In addition, to simplify the weight filter, the region above about 1kHz is designed to be flat. The RLB weight filter has a second order IIR filter structure and provides ITU-R documents with filter coefficients for 48 kHz data.

The result of passing through the weight filter is transformed as shown in the following equation in the mean-square energy module of FIG.

In this case, y _i is the input signal filtered by the pre-filter and the RLB filter modeling the head effect. T is the measurement period of the input signal, and z _i is the mean value of the filtered input signal. The weighted energy is added to the energy of each channel by applying the weight to each channel as shown in the following equation, and then converted into decibel by applying it to the logarithmic equation. The unit for the sound size obtained by the formula below is LKFS (loudness, K weighted, relative to nominal full scale).

In the equation, N is the number of channels and G is the weight for the channel.

In order to verify that the designed ITU-based audio sound size measurement method is correctly designed, a sound size measurement value of -3.01 LKFS should be output when 0dB, 1kHz sine waveform is input.

The study on the size of existing audio signal can be roughly classified into two kinds. The first is the development of an objective audio signal size measurement algorithm similar to the audio volume level audibly perceived by a person, such as ITU-R1770-1.

Secondly, in the past, the audio signal size was transmitted in an irregular manner, and accordingly, the volume of the audio file and the sound source that we listened to were different, so that the control of the audio signal size was automatically performed when audio files having different sizes were input .

In order to overcome the problem of audio signal size in each country, the audio signal size is measured based on ITU-1770-1 / 2, and the audio signal size normalization reference value and error range are presented based on this. Currently, Japan is actively engaged, but the rest of the country is still in its infancy and is only applied to parts such as commercials.

In other words, the contents included in the standardization and regulatory bill define the normalization mood, the error range, and the scope of application, but do not provide a method for meeting these standards. In other words, we have presented only the goals that need to be done and did not show the method.

Meanwhile, an audio gating method was added to the ITU-R audio signal size measurement method revised in March 2011. Audio Gating is a method for measuring the audio volume, except for low audio volume.

Audio volume measurement The block for gating is one cycle, overlapping 75% with neighboring blocks. Also, do not measure samples that do not satisfy the block size at the end of the file.

First, the mean square of a block unit is calculated according to the following equation.

In this case, z _ij denotes the mean square of the jth Gating block unit of the i-th input channel in the T interval. The audio volume of each gated block is calculated based on the same formula as the following.

When Gating is applied to each block, ITU-R 1770-2 measures LFKS for the signal with Gating applied only for signal of -70LKFS or more.

In this case, J _g is a set of gating block indices, and L _KG is a gated loudness in a T section. In the revised method, if the existing pre-filter and RLB filter are used in the same way, the method of verifying the accuracy of the algorithm is also the same.

In reference to the above description, contents included in the standardization and regulatory legislation so far define the normalization standard, the error range, and the scope of application, but do not clearly disclose the method for meeting the standard.

Accordingly, according to the first embodiment of the present invention to be described later, it is possible to control the size of the audio signal for the recording and the previously produced broadcast program to conform to the standard.

In addition, according to the second embodiment of the present invention to be described later, it is possible to control the size of the audio signal for the real-time / live-acquired broadcast program to conform to the standard.

In addition, according to the third embodiment of the present invention described below, the audio signal size can be controlled while minimizing auditory audio quality deterioration resulting from normalization of the audio signal size.

In addition, according to the fourth embodiment of the present invention to be described later, a new audio control function in a terminal (TV, smart phone) can be provided in consideration of the normalization of the audio signal size.

9 is a diagram for explaining an example of a structure of a broadcast system for recording and a previously produced broadcast program.

Referring to FIG. 9, the audio data acquired in the field is stored in the Ingest server, and the stored file is transferred to the editing system. In editing system, editing is done for each part such as well known video / sound effect, audio noise removal, video / audio synchronization.

The data edited by each part is finally processed in the comprehensive editing system, and the edited broadcasting program is sent out in the main control room. According to such a structure, the audio signal size regulation and the audio signal size normalization operation for the pre-programmed broadcast program can be performed in the editing system and the comprehensive editing system. Preferably, Since the editing system independently controls the audio data, it can be performed as an editing system post-processing operation.

10 is a diagram showing a first embodiment of a method of controlling the size of an audio signal.

In case of a previously recorded broadcast program file, the audio signal size must be normalized by analyzing the stored file. Accordingly, referring to FIG. 10, the demultiplexer demultiplexes the previously recorded broadcast program file to select audio data (S101).

Then, the normalization determination unit can determine whether the audio data is pre-normalized (S102). Here, the normalization means that the audio signal size is adjusted according to the standardized audio signal size standard as shown in FIG. 5 and normalized.

If pre-normalization has been performed on the audio data (S102: Y), the normalized audio data may be stored in the storage device (S103).

If no pre-normalization has been performed on the audio data (S102: N), the audio decoder can decode the audio data (S104). The audio signal size controller can perform normalization of the audio signal size using the decoded audio data (S105). Then, the audio encoder can encode the normalized audio data (S106).

On the other hand, the multiplexer can multiplex the encoded audio data with other data not selected in the demultiplexer (S107). Accordingly, the storage unit may store the audio data whose audio signal size is normalized (S103).

The data stored in the storage unit may be provided to the delivery room (S108).

Here, the concrete operation of the audio signal size controller will be described in detail with reference to FIGS. 11 to 12. FIG.

Meanwhile, the dotted line blocks shown in the figure, for example, steps S101, S104, S106, and S107 may be omitted depending on the format of the audio data and the like. For example, steps S104 and S106 may be omitted depending on whether or not the audio data is compressed.

According to the first embodiment of the present invention, in order to control the audio volume for recording and for converting a previously prepared broadcast program into an audio volume standard, Based on this, it is possible to measure and control the necessary audio volume according to the audio volume regulation.

11 is a diagram for explaining a first embodiment of a method for controlling the size of an audio signal. 12 is a diagram illustrating a basic structure of a peak value based loudness control ratio calculation for adjusting an audio signal size. Hereinafter, in describing Figs. 11 to 12, a detailed description of the parts described in Fig. 10 will be omitted.

Referring to FIG. 11, control information may be provided to control a recorded broadcast program.

Targeted audio signal sizes (Target LKFS) and audio signal size tolerance ranges set by regulations and legislation in various countries around the world can be provided first. In general, the US / Japan has a range of 24 LKFS (Target LKFS) +/- 2 dB (error range), and Europe has a range of 23 LKFS (Target LKFS) +/- 1 dB (error range).

The audio gating part is the first part mentioned in ITU-R 1770-2. It is measured by LKFS for each block by applying Overlap and shift method. It is regarded as having silence for low block LKFS, Method.

ATSC in the US is using the AC-3 audio system to store the "dialnorm" parameter in the metadata parameter. Dialnorm is intended to contain the auditory audio signal size for the anchor element, ie the size of the reference point or element's auditory audio signal.

The anchor element represents the standard audio signal size for the center of the current program, and the broadcasting program is finally balancing based on the anchor element. Also, the dialnorm stores the LKFS value, which can store -1 to -31 LKFS in variable space of 5 bits.

On the other hand, in order to measure the audio signal size based on ITU-R, two filters must be applied. Therefore, even if the measured LKFS and LKFS difference values are subtracted by the LKFS measurement equation, the audio signal size conversion value is extracted, so that the two filters are influenced.

In order to overcome this problem, according to the first embodiment of the present invention, it is possible to provide an algorithm for obtaining an audio signal size conversion weight factor according to a desired target LKFS by designing a method using a Peek value.

As described above, accurate loudness (LD) control ratios can not be obtained for the reason explained above only by the LKFS (original) and Target LKFS of the input audio.

Therefore, according to the first embodiment of the present invention, a peak-based control ratio can be calculated using a peeking method in order to obtain an LD control ratio considering two filters. The peeking method may be a method of obtaining a peeked LKFS by loudness control of an audio signal using a Peek-based control ratio. That is, the audio signal size controller receives the input audio data S105-1, Peek weight (ex.0.9) (S105-2), the target value LKFS (S105-3), and the LKFS error range 105-4, A loudness control ratio for controlling the size can be calculated (S105-5), and the LD control ratio can be calculated (S105-6). The LKFS of the input audio data calculated based on the input audio data, the Peek LKFS calculated by applying the Peek weight to the input audio data, and the target LKFS to calculate the wight factor (LD control ratio) can be calculated.

The new ratio is equal to diff1 / diff2 of Equation (7), where Ori LKFS is the LKFS value of the measured input audio signal and Peek LKFS is the scaling of the input audio signal using the initial Peek weight Value, Req LKFS is the target LKFS value, that is, Requested LKFS, which is the same as Target LKFS. The audio signal size controller can perform normalization by adjusting the size of the input audio signal using the calculated LD control ratio.

According to the first embodiment of the present invention, it is possible to control the size of the audio signal to be conformed to the standard for the recording and the previously produced broadcast program.

13 is a diagram showing an example of a structure of a real-time broadcasting system.

Referring to FIG. 13, the live broadcast system differs greatly from the broadcast broadcast system. The relay system does not include an ingest server and does not use a part-by-part editing system. Instead, in a live broadcasting system, the relay system performs these functions integrally.

The relay system carries out operations such as video / audio editing and effects, and performs control of the audio sound which is live through the mutual instruction and the sub-control room (comprehensive editing room) which manages the entire production of the program.

This coordinated broadcast program will be sent out from the main control room. In addition, live streaming data received via satellites are transmitted through the main control room by performing an additional operation such as work on audio sound and subtitle insertion in the subordinate control room (comprehensive editing room). Therefore, there are more variables to accurately control the audio volume of live broadcasts.

14 is a diagram showing a second embodiment of a method of an apparatus for controlling the size of an audio signal.

Referring to FIG. 14, in the live environment, a micro-acquired signal and a signal received by satellite (hereinafter, a live broadcast signal) can be considered as described above. The demultiplexer demultiplexes the live broadcast signal to select audio data (S201). Then, the audio decoder can decode the selected audio data (S203).

The audio signal size controller may perform normalization of the audio signal size using the decoded audio data (S206). Specifically, the audio signal size controller can analyze the size of the audio signal of the live audio data, and perform normalization by controlling the size of the live audio signal. Here, the audio signal size controller may perform normalization using the audio signal size control value manually input from the user (S205).

Then, the audio encoder can encode the audio data subjected to the normalization (S207). Then, the multiplexer can multiplex the encoded audio data with other data not selected in the demultiplexer (S208).

On the other hand, when the above-described data processing is performed, data may be provided to the transmission room (S209).

Here, the concrete operation of the audio signal size controller will be described in detail with reference to FIG.

Meanwhile, the dotted line blocks shown in the figure, for example, steps S201, S203, S205, S207, and S208 may be omitted depending on the format of audio data or the like. For example, if the input file is audio raw data, no audio decoding is required, and if an audio raw file is requested as the output, an audio encoding module is not required. When the signal is transmitted in streaming, the audio signal size control system decodes the audio signal into an audio signal if the audio data is a compressed bitstream after demuxing the file, and bypasses the audio decoding block if the audio data is raw data. The audio raw signal automatically adjusts the live audio signal according to the audio signal size criterion, and the adjusted signal is broadcasted through the transmission device by performing audio encoding and file formatting as necessary. Alternatively, an audio raw file can be output according to the request from the output.

FIG. 15 is a diagram for specifically explaining a second embodiment of a method for controlling the size of an audio signal. Hereinafter, in describing FIG. 15, a detailed description of the portion described with reference to FIG. 14 will be omitted.

Referring to FIG. 15, in the proposed system, unlike the conventional system, the three modes are possible in relation to the normalization of the audio signal size (S206). The first is Manual Loudness control mode, the second is Half automatic Loudness control mode, and the third is Automatic Loudness control mode. Each mode can operate independently, and it can operate in each mode and convert to another mode in the middle. The mode change part can be compensated by the module for the difference between the two modes by Mode Change Control.

The manual loudness control mode is a mode in which a person (for example, an audio signal editor) manually selects a weight for controlling the size of an audio signal input (for example, by using various buttons provided in the audio signal processor) And scales the input audio signal to control the audio signal size to match the target audio signal size. The Half Automatic Loudness control mode is the same as Manual Loudness control mode in that a manual selection of a weight for control is made by the user. In the Half Automatic Loudness control mode, information necessary for controlling the audio signal size (for example, And the size of the input audio signal) in order to provide the above-mentioned information so that the information can be used by a person. The Automatic Loudness control mode may be a mode that automatically controls the audio signal size to match the target audio signal size without manual control of the person.

On the other hand, in FIG. 15, the weights necessary for matching the target audio signal size (Target LKFS) with respect to the real-time input audio signal can be calculated through the above-described peeking method.

According to the second embodiment of the present invention, the size of the audio signal can be controlled to meet the standard for the real-time / live-acquired broadcast program.

16 is a diagram for explaining a method in which a live LD control step is added to the final stage of the first embodiment and the second embodiment. Referring to FIG. 16, a live LD control step may be further included in the final stage of the method according to the first and second embodiments of the present invention.

That is, according to the above description, the file / local broadcast program can be stored in the storage unit through the Local LD Control (S105) (S103) and used for transmission. In addition, according to the above description, the live broadcast program can be transmitted in real time through the Live LD Control (S206) and transmitted.

However, in case of a broadcasting station, Live LD Control (S210) may be further performed at the final stage in order to prepare for the regulation. That is, in the case of a broadcast station, even if a broadcast program erroneously inputted at the previous stage is delivered, a Live LD Control (S210) may be further provided so that it can be filtered at the final stage. In this case, the Live LD Control (S210) can use the Manual Loudness control mode, the Half automatic Loudness control mode, or the Automatic Loudness control mode. However, the Automatic Loudness control mode can be used so that it can be processed automatically at all times, preferably for 24 hours.

17 is a diagram showing a third embodiment of a method for compensating sound quality degradation due to size control of an audio signal.

As described above, the method of controlling the audio signal size can be variously performed according to the condition of the input data. However, if the audio signal size is adjusted to the target LKFS and the error range, the configuration of the audio signal may become flat.

This is an adverse effect of the normalization of the audio signal size. In order to achieve the purpose of audio signal size normalization, the adverse effects of normalization must be resolved so that the power of audio normalization and user satisfaction can be improved.

Accordingly, according to the third embodiment of the present invention, it is possible to further comprise an acoustic degradation compensation module for compensating for the above-mentioned adverse effect. That is, referring to FIG. 17, the demultiplexer demultiplexes existing broadcast program data or live broadcast program data to select audio data (S301).

Then, the normalization determination unit may determine whether the audio data is pre-normalized (S302).

If pre-normalization has been performed on the audio data (S302: Y), a subsequent procedure for the normalized audio data may be performed (S303).

If no pre-normalization has been performed on the audio data (S302: N), the audio decoder can decode the audio data (S304). Then, an editor control such as Live Audi Mixing & EQ may be performed (S305). The audio signal size controller may perform normalization of the audio signal size using the decoded audio data (S306).

Then, the auditory deterioration compensation module can compensate for the adverse effect of the normalization performed in the small signal size controller (S307). Then, the audio encoder may encode the audio data on which the auditory deterioration compensation has been performed (S308).

Then, the multiplexer can multiplex the encoded audio data with other data not selected in the demultiplexer (S309).

Meanwhile, the dotted line blocks shown in the figure, for example, steps S301, S304, S308, and S309 may be omitted depending on the format of the audio data. For example, steps S304 and S308 may be omitted depending on whether the audio data is compressed.

According to the third embodiment of the present invention, audio signal size can be controlled while minimizing auditory audio quality deterioration due to normalization of the audio signal size.

Meanwhile, the normalization of the audio signal size according to the above-described method invites a considerable change in the listening environment for the digital broadcasting consumer. In addition, since the size of the audio signal is normalized, newly required services / functions in the digital broadcasting terminal can be generated. That is, the digital broadcast terminal can provide functions related to broadcast audio volume.

18 is a diagram showing a fourth embodiment of a method for controlling the size of an audio signal in a terminal. Hereinafter, with reference to FIG. 18, a detailed description of the portion described in FIG. 17 (processing portions (S301 to S3010) related to the transmission of the normalized audio signal will be omitted.

Referring to FIG. 18, the terminal receives a normalized audio signal (S401), processes the received audio signal (S402), and outputs the processed audio signal (S403). Here, the audio signal processing (S402), for example, can be controlled in a customized manner. That is, in digital broadcasting, information about broadcasting is provided to the user, and when the user continuously uses the terminal, usage information of the user is accumulated. Based on this information, user information analysis is performed, and a customized audio sound service can be provided to the user. Also, the user information service based on broadcasting information can be directly applied according to the user setting information.

FIG. 19 is a flowchart specifically illustrating a method of controlling the audio signal size of the audio signal size control apparatus according to the first embodiment of the present invention. Referring to FIG. 19, an audio signal can be received first (s501). The audio signal input here may be, for example, an audio signal according to an operation (an operation that can be omitted) such as demultiplexing and decoding shown in FIGS. 10 to 12. Such an audio signal may have various waveforms, and may be, for example, an audio signal having a waveform of a waveform shown in the front end of FIG. 5 (i.e., before being normalized).

In this case, the audio signal size measuring unit may measure LKFS (Original LKFS) of the input audio signal using the audio signal size measuring method described in FIGS. 6 to 8 (S503).

Also, the audio signal size measuring unit may measure the initial peak LKFS (S502). The initial Peek LKFS can be measured by scaling the input audio signal using a predetermined initial Peek weight and measuring the LKFS based on the scaled audio signal.

The predetermined initial Peek weight may be provided in the form of control information to a broadcast signal including an audio signal and a video signal. Or as a pre-stored value at the time of design of the audio signal size control device. Or may be provided as input from a user.

On the other hand, the weight calculation unit calculates the weight LKFS (Original LKFS) (S504) of the measured input audio signal, the measured initial Peek LKFS (S502), the target LKFS (Target LKFS) S503), the audio signal size control ratio can be calculated (S506). Specifically, the weight calculation unit may calculate an audio signal size control ratio using Equation (7) below.

Here, the original LKFS is the measured LKFS value of the input audio signal, Peek LKFS is the measured value based on the scaled input audio signal using the initial Peek weight, and the target LKFS is the target LKFS value, diff1 is the difference value between the original LKFS and Peek LKFS, and diff2 is the difference value between the original LKFS and the target LKFS. On the other hand, the audio signal size control ratio may be diff1 / diff2.

Then, the weight calculation unit may calculate a new peak weight (S507) by applying the calculated audio signal size control ratio to the following expression (8).

Here, new_Peek_weight denotes a new Peek weight, previous_Peek_weight denotes a Peek weight used before calculating new_Peek_weight, and new_weight denotes a weight calculated using an audio signal size control ratio in Equation (8). For example, according to Equations (7) to (8) described above, a new Peek weight can be calculated by multiplying an initial Peek weight by new weight at first (S505: Y).

In Equation (8), when the difference between the original LKFS and the Peek LKFS is smaller than the difference between the original LKFS and the target LKFS, the new Peek weight is calculated by decreasing the previous peak weight and the difference between the original LKFS and the Peek LKFS If the difference between the original LKFS and the target LKFS is larger than the difference between the original LKFS and the target LKFS, the new peak weights can be calculated by increasing the previous peak weights.

In Equation (8), the weight for decreasing is 0.9, and the weight for increasing is 1.1, but various weight values can be used instead of the weight. For example, for a finer audio signal size adjustment, the weight for the reduction may be 0.99, and the weight for the increase may be 1.01.

Here, the target value LKFS (Target LKFS) can be changed according to the target value LKFS (Target LKFS) defined by the regulations and legislation of various countries in the world. As an example, the target value LKFS (Target LKFS) may be? 24 LKFS, as shown at the end of FIG. 5 (i.e., after normalization). This target value LKFS (Target LKFS) may be provided in the form of control information to a broadcast signal including an audio signal and a video signal. Or as a pre-stored value at the time of design of the audio signal size control device. Or may be provided as input from a user.

On the other hand, the audio signal size control unit can control the audio signal size using the new Peek weight calculated by the above operation. Specifically, the audio signal size control unit may control the size of the audio signal by scaling the input audio signal S501 using the calculated new peak weight (New Peek weight) (S508).

The audio signal size measuring unit may measure LKFS (New Peek LKFS) of the audio signal S508 whose audio signal size is controlled according to a new Peek weight (S509).

Meanwhile, the audio signal size controller may calculate the LKFS error by comparing the target value LKFS (Target LKFS) (S504) with the measured new Peek LKFS (S509) (S511).

Then, the audio signal size controller may compare the LKFS error D with a predetermined error range T (S512). For example, if the target value LKFS (Target LKFS) and the audio signal size error range are 24 LKFS (Target LKFS) +/- 2 dB (error range), the difference between the target value LKFS (Target LKFS) and the new Peek LKFS It is possible to judge whether it is larger or smaller than the error range. The predetermined error range (LKFS error range) (S510) may be provided in the form of control information to a broadcast signal including an audio signal and a video signal. Or as a pre-stored value at the time of design of the audio signal size control device. Or may be provided as input from a user.

If it is small (S513: Y), the audio signal size control unit may output an audio signal whose audio signal size is controlled according to a new peak weight (New Peek weight).

If it is larger (S513: N), the audio signal size control section can control to repeat the above-described control operation. When the above-described control operation is repeated, the weight calculation unit calculates the target value LKFS (Target LKFS) (S504), the measured new Peek LKFS (S505: N), the measured new Peek LKFS A new audio signal size control ratio can be calculated using the measured original LKFS (S503) (S506). In this case, the weight calculation unit can calculate the loudness control ratio using Equation (7). In addition, the weight calculation unit may calculate a new peak weight (S507) by applying the calculated audio signal size control ratio to Equation (8). That is, the above-described operation can be repeated until the size of the audio signal satisfies the target value LKFS (Target LKFS) and the error range.

Meanwhile, the input audio signal S501 according to the first embodiment of the present invention may be an audio signal for a pre-programmed program, and may be an audio signal for a program from the start to the end. Accordingly, according to the first embodiment of the present invention, the audio signal size can be controlled based on the audio signal size (Original LKFS) of the audio signal from the start to the end of the broadcast program.

On the other hand, an encoding operation, a multiplexing operation (not shown), and the like shown in Figs. 10 to 12 may be performed on the output audio signal S513.

The apparatus or method for controlling the audio signal size according to the first embodiment of the present invention may be provided or performed on the maker side of producing the audio signal or on the supplier side that supplies the produced audio signal. Alternatively, the audio signal size control apparatus or method according to the first embodiment of the present invention may be provided or performed on a user side (for example, a portable multi-device such as an MP3 player) receiving and outputting an audio signal.

According to the first embodiment of the present invention described above, it is possible to automatically control the size of the audio signal for the recording and the previously produced broadcast program to conform to the standard.

20 is a view for explaining an audio signal size measurement method to which the audio gating method described in ITU-R 1770-2 is added. Here, the audio Gating method measures the LKFS for the Gate Block 2 by measuring the LKFS for the Gate Block 1, the Overlap and Shift method as shown in FIG. 20, and repeats the Overlap and shift method for the Gate Block 2, And if the measured LKFS of the gate block is less than or equal to the threshold LKFS (-70 LKFS in ITU-R 1770-2), silence processing is performed to perform audio signal size measurement for the audio signal to which the gating is applied.

As for the above-mentioned Gate Block, in ITU-R 1770-2, the Gate Block has a gate size of 0.4s and has a structure of 75% Overlap.

On the other hand, in the real-time / live environment, the audio signal is acquired for each gate block, and the LKFS for each gate block is measured using Equations 4 to 5 described above. A new peak weight (new peak weight) for controlling can be calculated using the method of Fig. 19 described above. However, if the audio signal size is controlled for each gate block by using a new peak weight calculated for each gate block, a discontinuous sound may be generated due to a weight difference between neighboring gate blocks. .

In order to solve such a problem, the audio signal size control method according to the fifth embodiment of the present invention can perform the following processing.

21 is a view for explaining a gate handover in order to explain an audio signal size control method according to a fifth embodiment of the present invention. Referring to FIG. 21, the gate size of the non-overlapping area of the gate block may be 4800 samples, for example. In addition, when using a codec such as AAC, AC-3, etc., one frame size for determining the size of data to be received at one time may be 1024 samples. In this case, a gate handover may occur where one frame spans two gate blocks.

22 is a view for explaining an audio signal size control method according to the fifth embodiment of the present invention. Referring to FIG. 22, an audio signal size control method according to a fifth exemplary embodiment of the present invention can control an audio signal size by interpolating a gate weight from a frame in which a gate handover occurs, have. Here, the gate weight may be a new peak weight calculated using the method of FIG. 19 for each Gate Block.

According to the fifth embodiment of the present invention, gate delay due to interpolation of gate weights does not occur. That is, at the time of receiving data in a frame in which a gate handover occurs, gate weights for two gate blocks over which a gate handover occurs can be calculated in advance This is because the gate weights for two gate blocks calculated in advance can be used to interpolate the gate weights without delaying from the frame timing at which gate hand over occurs.

Meanwhile, according to the fifth embodiment of the present invention, various interpolation methods may be used to interpolate the gate weights. As an example, this linear interpolation can be used. This will be described in detail with reference to FIG.

 23 is a view for explaining linear interpolation, which is an example of interpolation according to the fifth embodiment of the present invention. Referring to FIG. 23, linear interpolation such as the following equation can be used.

In Equation (9), W _G1 is the gate weight of Gate Block 1, W _G2 is the gate weight of Gate Block 2, i is the number of gate weights to interpolate, and InterFrame is the number of frames from the interpolation start frame to the type frame.

For example, when the number of InterFrame is 3 and applied to Equation (9), gate weights (weighted values W ₁ and W 2 shown in red) to be applied to two frames can be calculated as shown in FIG. have. In other words, the number of interframes of the gate weights can be variably controlled by selectively adjusting the number of InterFrames.

Meanwhile, the gate weighting interpolation method according to the fifth embodiment of the present invention can be applied to a method of controlling an audio signal size using gate weights. For example, it can be applied to a previously recorded broadcast program to control the size of an audio signal, and it can be applied to a live broadcast program to control an audio signal size.

In addition, an apparatus or method for controlling an audio signal size according to a fifth embodiment of the present invention may be provided or performed on a maker side that manufactures an audio signal, or on a provider side that supplies a manufactured audio signal. Alternatively, the audio signal size control apparatus or method according to the fifth embodiment of the present invention may be provided or performed on a user side (e.g., a portable multi-device such as an MP3 player) receiving and outputting an audio signal.

According to the fifth embodiment of the present invention, gate weights are interpolated from a frame in which a gate handover occurs, so that gate delay due to interpolation of gate weights can be prevented.

In addition, the number of times the gate weights are interpolated can be variably controlled.

24 to 26 are diagrams comparing the waveforms of the input audio signal and the normalized audio signal.

Fig. 24 (a) is a waveform of an input audio signal of pop, and Fig. 24 (b) is a diagram of waveforms of a normalized audio signal of pop. Referring to FIG. 24, the size of the input audio signal is -22.23 LKFS, but the normalization operation is performed so that the normalized audio signal has a size of -22.72 LKFS and is normalized within the target audio signal size and error range. .

Fig. 25 (a) shows a waveform of the input audio signal of Kpop, and Fig. 25 (b) shows a waveform of the normalized audio signal of Kpop. Referring to FIG. 25, the size of the input audio signal is -8.9 LKFS. However, the normalization operation described above is performed, and the size of the normalized audio signal is -23.28 LKFS, which is normalized within the target audio signal size and error range. .

Fig. 26 (a) shows a waveform of an input audio signal of classic, and Fig. 26 (b) shows a waveform of a normalized normalized audio signal. Referring to FIG. 26, the size of the input audio signal is -26 LKFS, but the normalization operation is performed so that the normalized audio signal has a size of -25.34 LKFS and is normalized within the target audio signal size and error range. .

Meanwhile, the method according to various embodiments of the present invention may be stored in a computer-readable recording medium. The computer-readable recording medium may be a ROM, a RAM, CD-ROMs, magnetic tapes, floppy disks, optical data storage devices, and the like, as well as carrier waves (e.g., transmission over the Internet).

The computer readable recording medium may be distributed over a networked computer system so that computer readable code can be stored and executed in a distributed manner. And, functional programs, codes and code segments for implementing the above method can be easily inferred by programmers of the technical field to which the present invention belongs.

While the present invention has been particularly shown and described with reference to exemplary embodiments thereof, it is to be understood that the invention is not limited to the disclosed exemplary embodiments, but, on the contrary, It should be understood that various modifications may be made by those skilled in the art without departing from the spirit and scope of the present invention.

Claims (10)

A method for controlling an audio signal size using gate blocks to which an input audio signal is applied,
Measuring a first audio signal size that is a magnitude of an input audio signal to a first gate block;
Measuring a magnitude of a second audio signal that is a magnitude of an input audio signal to a second gate block;
Calculating a first gate weight based on the size of the first audio signal and calculating a second gate weight based on the size of the second audio signal;
Dividing the first gate block and the second gate block into frame units;
Detecting a frame in which a gate handover occurs in the first gate block and the second gate block; And
Interpolating the segmented frame using the first gate weight and the second gate weight;
/ RTI > of claim 1, wherein the audio signal size control method comprises:
The method according to claim 1,
Wherein the second gate block comprises:
And overlapping with the first gate block by a predefined ratio.
The method according to claim 1,
The first gate block and the second gate block may include a first gate block,
Comprising at least one frame,
Wherein a data size that can be received at a time is determined by each frame.
The method according to claim 1,
Wherein the number of frames to be interpolated by using the first gate weight and the second gate weight among the frames to be divided is variably adjusted.
5. The method of claim 4,
Wherein the interpolation of the first gate weight and the second gate weight is performed by the following equation:

Wherein W _G1 also refers to the number of frames to the first gate weights, W _G2 is the second gate weights, i is the sequence number, the InterFrame of the interpolation frame type from the start frame to be interpolated frame.
An apparatus for controlling an audio signal size using gate blocks to which an input audio signal is applied,
An audio signal size measuring unit measuring the size of the first audio signal as the size of the input audio signal to the first gate block and measuring the size of the second audio signal as the size of the input audio signal to the second gate block;
A weight calculation unit for calculating a first gate weight based on the size of the first audio signal and a second gate weight based on a size of the second audio signal;
A detector for dividing the first gate block and the second gate block into frame units; A detector for detecting a frame in which a gate handover has occurred in the first gate block and the second gate block; And
An audio signal size controller for interpolating the divided frame using the first gate weight and the second gate weight;
And an audio signal size control device.
The method according to claim 6,
Wherein the second gate block comprises:
And overlapping with the first gate block by a predefined ratio.
The method according to claim 6,
The first gate block and the second gate block may include a first gate block,
Comprising at least one frame,
And a data size that can be received at one time is determined by each frame.
The method according to claim 6,
Wherein the number of frames to be interpolated into the first gate weight and the second gate weight among the frames to be divided is variably adjusted.
10. The method of claim 9,
Wherein the interpolation of the first gate weight and the second gate weight is performed by the following equation:

Wherein W _G1 also refers to the number of frames to the first gate weights, W _G2 is the second gate weights, i is the sequence number, the InterFrame of the interpolation frame type from the start frame to be interpolated frame.

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