WO2007049222A1 - Adaptive volume control for a speech reproduction system - Google Patents

Adaptive volume control for a speech reproduction system Download PDF

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Publication number
WO2007049222A1
WO2007049222A1 PCT/IB2006/053901 IB2006053901W WO2007049222A1 WO 2007049222 A1 WO2007049222 A1 WO 2007049222A1 IB 2006053901 W IB2006053901 W IB 2006053901W WO 2007049222 A1 WO2007049222 A1 WO 2007049222A1
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Prior art keywords
gain
audio input
signal
input frame
audio
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PCT/IB2006/053901
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French (fr)
Inventor
Vincent Demanet
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Koninklijke Philips Electronics N.V.
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Publication of WO2007049222A1 publication Critical patent/WO2007049222A1/en

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Classifications

    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G3/00—Gain control in amplifiers or frequency changers
    • H03G3/002—Control of digital or coded signals
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L15/00—Speech recognition
    • G10L15/02—Feature extraction for speech recognition; Selection of recognition unit
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G3/00—Gain control in amplifiers or frequency changers
    • H03G3/20—Automatic control
    • H03G3/30—Automatic control in amplifiers having semiconductor devices
    • H03G3/3089—Control of digital or coded signals
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L15/00—Speech recognition
    • G10L15/20—Speech recognition techniques specially adapted for robustness in adverse environments, e.g. in noise, of stress induced speech
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/78—Detection of presence or absence of voice signals

Definitions

  • the invention relates generally to adaptive volume control for a speech reproduction system.
  • a sound reproduction system for use in, for example, GSM mobile telecommunications applications includes an output transducer, often called a loudspeaker, and an input for an audio signal.
  • the loudspeaker produces sound pressure waves in response to the audio input signal which is representative of a desired sound pressure wave.
  • DSP digital signal processor
  • An algorithm is used to apply a varying gain at the amplifier according to the level of the audio input signal.
  • one of the principal disadvantages of this kind of processing is its sensitivity to background noise included in the audio input signal.
  • the noise level can be substantially increased during a pause in the speech and there will be level variations of the noise depending on the speech level, also known as a "pumping" effect of the noise. Furthermore, careful tuning is required in order to avoid boosting the noise more than the speech level and to avoid modulating the background noise level with the speech.
  • US Patent No. 5,485,522 describes a system for adaptively reducing noise in frames of digitized audio signals which may include speech and background noise. Initially, it is determined whether a current audio frame contains speech information and an attenuation value determined for the previous frame is modified for application to the current frame, the modification being based upon an estimate of the noise in the current frame. Thus, the described system acts as a noise suppressor.
  • the gain is different between a speech frame and a subsequent noise frame (in the sense that, say, little or no attenuation is applied to a speech frame and a high level of attenuation is applied to a noise frame so as to suppress the noise)
  • the above-mentioned "pumping" effect occurs, which is considered to be unacceptable in many applications.
  • the input signal is divided by an attenuation factor that is always greater than 1, the output signal level is always smaller than the input signal level. When false detection occurs then the speech could also be attenuated like the noise.
  • an automatic gain control system for a sound reproduction apparatus having an input means for receiving an audio signal comprising a speech signal and noise, said audio signal being in the form of audio input frames, the system comprising means for receiving an audio input frame and data indicating that said audio input frame includes a speech signal, and means for determining a gain value representative of a gain to be applied to said audio input frame, wherein said gain value is updated relative to the gain value representative of a gain applied to a previous audio input frame to take into account the signal level of said audio input frame if said audio input frame, includes a speech signal, and, otherwise, said gain value is held constant at the gain value representative of the gain applied to a previous audio input frame.
  • a method for performing automatic gain control in a sound reproduction apparatus having an input means for receiving an audio signal comprising a speech signal and noise, said audio signal being in the form of audio input frames, the method comprising receiving an audio input frame and data indicating that said audio input frame includes a speech signal, and determining a gain value representative of a gain to be applied to said audio input frame, wherein said gain value is updated relative to a gain value representative of the gain applied to a previous audio input signal to take into account the signal level of said audio input frame if said audio input frame includes a speech signal, and, otherwise, said gain value is held constant at the gain value representative of the gain applied to a previous audio input frame.
  • clipping means are provided for clipping the audio signal at a predetermined amplitude (or clipping level), so as to ensure linear operation of the analogue amplifier employed in the sound reproduction apparatus.
  • the sound reproduction apparatus beneficially includes an input buffer having a maximum peak level, and the gain value is preferably calculated as the clipping level divided by the maximum peak level of the input buffer.
  • means are provided to limit the maximum resultant gain to be applied to the audio input frame to a predefined level.
  • ANL Automatic Volume Leveler
  • means are provided for smoothing the gain value if the input audio frame includes a sound signal. This has the advantage of avoiding too much signal distortion due to local waveform discontinuities at block (i.e. audio frame) edges.
  • Figure 1 is a schematic block diagram illustrating a sound reproduction system including a loudspeaker and a digital signal processor;
  • Figure 2 is a schematic block diagram illustrating the principal components of a volume control system according to an exemplary embodiment of the present invention
  • Figure 3 is a schematic flow diagram illustrating the principal steps of a volume control method according to an exemplary embodiment of the present invention.
  • a sound reproduction system can, for instance, be a hands-free loudspeaker cellular radiotelephone for use in an automobile.
  • speech signals received from a far end i.e. from a distant party, are transmitted from a cellular base station (not shown), received by the transceiver of the cellular phone (not shown), and applied to the input 1 for an incoming far end signal as an input waveform.
  • a cellular base station not shown
  • the transceiver of the cellular phone not shown
  • the system comprises an analog-to-digital (A/D) converter (not shown) to generate a digital far end signal which is then fed into input 1.
  • A/D analog-to-digital
  • the waveform is applied in a digital format at input 1 of a digital signal processor DSP 2, which is connected to (or comprises) a digital output 3.
  • the digital signal output is fed to a digital-to-analog (D/A) converter 4 and converted thereby to an analog format, following which it is amplified by an amplifier 5 for use by the loudspeaker 6.
  • An output sound pressure wave representative of the speech of the distant party is emitted by the loudspeaker 6. Accordingly, the radiotelephone user hears sound pressure waveforms which are representative of the speech of the distant party.
  • the input waveform does not just contain speech signals, but also background noise.
  • ADL Automatic Volume Leveler
  • a processing block is, for example, described in International Patent Application No. WO2005/004114.
  • the present invention provides a modified volume control system which is triggered by an input received from a speech detector in order to achieve the above-mentioned object.
  • a digital audio input frame is applied to a speech/noise detector 10.
  • speech/noise detector 10 There are many different types of speech detector known to a person skilled in the art (including that used in the arrangement of US Patent No. 5,485,522), and the output from any one of a number of them would be suitable for use in the present invention. The present invention is therefore not necessarily intended to be limited in this regard.
  • the digital audio input frame is also applied to a gain update module 12 where the gain to be applied to the audio input frame is updated (or not) relative to the gain applied to the previous audio input frame, according to the output of the speech detector 10. Once the value of the gain to be applied has been determined, it is applied at module 14. Finally, the signal to which the gain has been applied is fed to a limiter (or clipper) 16, which clips the signal at a certain amplitude, to ensure linear operation of the analog amplifier 5 (after D/A conversion).
  • the limiter 16 may comprise a hard clipper, which simply clips the signal above the clipping level, or a soft clipper may be used, which clips the signal above a clipping level but also attenuates the signal at a level close to the clipping level.
  • Steps 30 to 40 are performed by the gain update module 12.
  • the maximum peak level of the input buffer i.e. the maximum input level ML of the frame
  • the two main aims and therefore contributing factors are: - First it is there to normalize the speech, e.g. during a conference call, the audio level of the participants can be quite different, the AVL or AGC (stands for Automatic Gain Controller) will tend to normalize them to same level. Then the amplitude output will be kept almost constant.
  • the second goal is to optimize the use of the amplifier inside, for example, a mobile phone.
  • the amplifier and loudspeaker are limited in power due to voltage and size issues.
  • the gain value calculated at step 32 is smoothed at a step 36 to avoid too much signal distortion due to local waveform discontinuities at block edges.
  • the gain to apply GaintoApply at step 38 is determined as being equal to the smoothed gain value.
  • the gain to apply is applied to a limiter, at step 40, and the resultant gain is applied to the input signal Pcm samp i e at module 14.
  • the input audio frame may contain speech and noise, in which case, the gain value is updated and applied as described above, but the applied gain is still lower than that which is applied by conventional compressor systems due to the limiter.
  • the present invention provides a volume control module that is triggered by the output of a speech/noise detector.
  • the gain update will only be performed when a speech is detected in a frame.
  • no gain update is performed and the gain remains constant. Because the same gain value is used for a noise frame as was used for a previous speech frame, the so-called "pumping" effect referred to above is avoided.
  • the input signal Pcm samp i e is multiplied by ClipLevel/ML, whereby the clipping level is typically OdB (full scale) or -3dB, so the gain will, in most cases, be greater than 1, the output level will, in most cases, be greater than the input level, and there will be no noise attenuation, in contrast to the system described in US Patent No. 5, 485,522. Still further, the output from any one of a number of speech detectors may be used for the present invention.

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  • Engineering & Computer Science (AREA)
  • Computer Vision & Pattern Recognition (AREA)
  • Computational Linguistics (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Control Of Amplification And Gain Control (AREA)

Abstract

The invention relates to an adaptive value control in a speech reproduction system, wherein a speech detector determines if an audio input frame contains a speech signal. If the frame contains a speech signal, the gain to be applied to that frame is updated relative to the gain applied to a previous frame to take into account the signal level. If not, i.e. the frame contains only noise, the gain applied to the frame remains constant at that applied to the previous frame.

Description

ADAPTIVE VOLUME CONTROL FOR A SPEECH REPRODUCTION SYSTEM
FIELD OF THE INVENTION
The invention relates generally to adaptive volume control for a speech reproduction system.
BACKGROUND OF THE INVENTION
A sound reproduction system for use in, for example, GSM mobile telecommunications applications, includes an output transducer, often called a loudspeaker, and an input for an audio signal. The loudspeaker produces sound pressure waves in response to the audio input signal which is representative of a desired sound pressure wave. It is well known to employ automatic gain control in a digital signal processor (DSP) in order to hold the output volume constant. An algorithm is used to apply a varying gain at the amplifier according to the level of the audio input signal. However, one of the principal disadvantages of this kind of processing is its sensitivity to background noise included in the audio input signal. The noise level can be substantially increased during a pause in the speech and there will be level variations of the noise depending on the speech level, also known as a "pumping" effect of the noise. Furthermore, careful tuning is required in order to avoid boosting the noise more than the speech level and to avoid modulating the background noise level with the speech.
Intelligibility of the sound as perceived by the listener is very important, especially in noisy environments, and a number of attempts have been made to increase the intelligibility of sound produced by a sound reproduction system. US Patent No. 5,485,522 describes a system for adaptively reducing noise in frames of digitized audio signals which may include speech and background noise. Initially, it is determined whether a current audio frame contains speech information and an attenuation value determined for the previous frame is modified for application to the current frame, the modification being based upon an estimate of the noise in the current frame. Thus, the described system acts as a noise suppressor. However, because the gain is different between a speech frame and a subsequent noise frame (in the sense that, say, little or no attenuation is applied to a speech frame and a high level of attenuation is applied to a noise frame so as to suppress the noise), the above-mentioned "pumping" effect occurs, which is considered to be unacceptable in many applications. Furthermore, because the input signal is divided by an attenuation factor that is always greater than 1, the output signal level is always smaller than the input signal level. When false detection occurs then the speech could also be attenuated like the noise.
SUMMARY OF THE INVENTION It is therefore an object of the present invention to provide an improved volume control system and method for use in sound reproduction, which improves intelligibility of the speech and reduces the effects of background noise.
In accordance with the present invention, there is provided an automatic gain control system for a sound reproduction apparatus having an input means for receiving an audio signal comprising a speech signal and noise, said audio signal being in the form of audio input frames, the system comprising means for receiving an audio input frame and data indicating that said audio input frame includes a speech signal, and means for determining a gain value representative of a gain to be applied to said audio input frame, wherein said gain value is updated relative to the gain value representative of a gain applied to a previous audio input frame to take into account the signal level of said audio input frame if said audio input frame, includes a speech signal, and, otherwise, said gain value is held constant at the gain value representative of the gain applied to a previous audio input frame.
Also in accordance with the present invention, there is provided a method for performing automatic gain control in a sound reproduction apparatus having an input means for receiving an audio signal comprising a speech signal and noise, said audio signal being in the form of audio input frames, the method comprising receiving an audio input frame and data indicating that said audio input frame includes a speech signal, and determining a gain value representative of a gain to be applied to said audio input frame, wherein said gain value is updated relative to a gain value representative of the gain applied to a previous audio input signal to take into account the signal level of said audio input frame if said audio input frame includes a speech signal, and, otherwise, said gain value is held constant at the gain value representative of the gain applied to a previous audio input frame.
Thus, because the gain applied to an audio input frame having no speech signal, i.e. just noise, is the same as that applied to the last frame containing a speech signal, the above- mentioned "pumping" effect is avoided, because there is no noise attenuation as such.
Preferably, clipping means are provided for clipping the audio signal at a predetermined amplitude (or clipping level), so as to ensure linear operation of the analogue amplifier employed in the sound reproduction apparatus. The sound reproduction apparatus beneficially includes an input buffer having a maximum peak level, and the gain value is preferably calculated as the clipping level divided by the maximum peak level of the input buffer. In a preferred embodiment, means are provided to limit the maximum resultant gain to be applied to the audio input frame to a predefined level. Thus, even when an audio input frame includes speech and noise and the gain is applied to both, this gain will not be as high as in conventional compression systems. As a result, the system is more robust against false detection as it can never be worse than a conventional Automatic Volume Leveler (AVL).
Preferably, means are provided for smoothing the gain value if the input audio frame includes a sound signal. This has the advantage of avoiding too much signal distortion due to local waveform discontinuities at block (i.e. audio frame) edges.
BRIEF DESCRIPTION OF THE DRAWINGS
These and other aspects of the present invention will be apparent from, and elucidated with reference to, the embodiments described herein.
Embodiments of the present invention will now be described by way of examples only and with reference to the accompanying drawings, in which:
Figure 1 is a schematic block diagram illustrating a sound reproduction system including a loudspeaker and a digital signal processor;
Figure 2 is a schematic block diagram illustrating the principal components of a volume control system according to an exemplary embodiment of the present invention; and Figure 3 is a schematic flow diagram illustrating the principal steps of a volume control method according to an exemplary embodiment of the present invention.
DETAILED DESCRIPTION OF THE INVENTION
Referring to Figure 1 of the drawings, there is illustrated schematically, a sound reproduction system. Such a system can, for instance, be a hands-free loudspeaker cellular radiotelephone for use in an automobile. When implemented as a hands-free cellular telephone, speech signals received from a far end, i.e. from a distant party, are transmitted from a cellular base station (not shown), received by the transceiver of the cellular phone (not shown), and applied to the input 1 for an incoming far end signal as an input waveform. In this example, it is assumed that the transmission back and forth between the system, in this example, a telephone system, and the far end is in digital form. If the original signals are in analog form, the system comprises an analog-to-digital (A/D) converter (not shown) to generate a digital far end signal which is then fed into input 1. As shown in Figure 1, the waveform is applied in a digital format at input 1 of a digital signal processor DSP 2, which is connected to (or comprises) a digital output 3. The digital signal output is fed to a digital-to-analog (D/A) converter 4 and converted thereby to an analog format, following which it is amplified by an amplifier 5 for use by the loudspeaker 6. An output sound pressure wave representative of the speech of the distant party is emitted by the loudspeaker 6. Accordingly, the radiotelephone user hears sound pressure waveforms which are representative of the speech of the distant party. However, the input waveform does not just contain speech signals, but also background noise.
It is known to provide an Automatic Volume Leveler (AVL), which is a signal- dependent processing block for keeping the volume of the incoming signal at an approximately constant level, wherein an updated gain is applied to each frame of the audio input signal according to the signal level thereof. Such a processing block is, for example, described in International Patent Application No. WO2005/004114. The present invention provides a modified volume control system which is triggered by an input received from a speech detector in order to achieve the above-mentioned object. Thus, referring to Figure 2 of the drawings, in general, a digital audio input frame is applied to a speech/noise detector 10. There are many different types of speech detector known to a person skilled in the art (including that used in the arrangement of US Patent No. 5,485,522), and the output from any one of a number of them would be suitable for use in the present invention. The present invention is therefore not necessarily intended to be limited in this regard.
The digital audio input frame is also applied to a gain update module 12 where the gain to be applied to the audio input frame is updated (or not) relative to the gain applied to the previous audio input frame, according to the output of the speech detector 10. Once the value of the gain to be applied has been determined, it is applied at module 14. Finally, the signal to which the gain has been applied is fed to a limiter (or clipper) 16, which clips the signal at a certain amplitude, to ensure linear operation of the analog amplifier 5 (after D/A conversion). The limiter 16 may comprise a hard clipper, which simply clips the signal above the clipping level, or a soft clipper may be used, which clips the signal above a clipping level but also attenuates the signal at a level close to the clipping level. Using a soft clipper restores, to a certain extent, the dynamic behavior of the signal, which reduces distortion and increases intelligibility. The speech detection function and the automatic volume control functions are performed by the digital signal processor 2 of Figure 1 in accordance to the schematic flow diagram illustrated in Figure 3. Steps 30 to 40 are performed by the gain update module 12. First, the maximum peak level of the input buffer (i.e. the maximum input level ML of the frame) is determined, at a step 30, and then a gain value Gain is calculated at a step 32, where Gain = ClipLevel/ML, where ClipLevel is the clipping level. The purpose of this gain value, and the considerations in its determination relative to automatic volume control, are well known to a person skilled in the art. The two main aims and therefore contributing factors are: - First it is there to normalize the speech, e.g. during a conference call, the audio level of the participants can be quite different, the AVL or AGC (stands for Automatic Gain Controller) will tend to normalize them to same level. Then the amplitude output will be kept almost constant.
The second goal is to optimize the use of the amplifier inside, for example, a mobile phone. The amplifier and loudspeaker are limited in power due to voltage and size issues.
Thus, normalizing the amplitude of their input signal tends to push them into their maximum performances, so it provides the user a more comfortable loudness, especially in speaker phone mode.
Next, it is determined at a step 34 whether or not the input signal is a speech frame, depending on the output of the speech detector 10. If so, the gain value calculated at step 32 is smoothed at a step 36 to avoid too much signal distortion due to local waveform discontinuities at block edges. Thus, the gain to apply GaintoApply at step 38 is determined as being equal to the smoothed gain value. Next, the gain to apply is applied to a limiter, at step 40, and the resultant gain is applied to the input signal Pcmsampie at module 14. Of course, the input audio frame may contain speech and noise, in which case, the gain value is updated and applied as described above, but the applied gain is still lower than that which is applied by conventional compressor systems due to the limiter.
If, on the other hand, the input frame is determined at step 34 to be a noise frame, the gain value applied to the previous input frame is applied to the current frame. Thus, the present invention provides a volume control module that is triggered by the output of a speech/noise detector. The gain update will only be performed when a speech is detected in a frame. During a noise frame, no gain update is performed and the gain remains constant. Because the same gain value is used for a noise frame as was used for a previous speech frame, the so-called "pumping" effect referred to above is avoided. Furthermore, the input signal Pcmsampie is multiplied by ClipLevel/ML, whereby the clipping level is typically OdB (full scale) or -3dB, so the gain will, in most cases, be greater than 1, the output level will, in most cases, be greater than the input level, and there will be no noise attenuation, in contrast to the system described in US Patent No. 5, 485,522. Still further, the output from any one of a number of speech detectors may be used for the present invention.
It should be noted that the above-mentioned embodiments illustrate rather than limit the invention, and that those skilled in the art will be capable of designing many alternative embodiments without departing from the scope of the invention as defined by the appended claims. In the claims, any reference signs placed in parentheses shall not be construed as limiting the claims. The word "comprising" and "comprises", and the like, does not exclude the presence of elements or steps other than those listed in any claim or the specification as a whole. The singular reference of an element does not exclude the plural reference of such elements and vice- versa. The invention may be implemented by means of hardware comprising several distinct elements, and by means of a suitably programmed computer. In a device claim enumerating several means, several of these means may be embodied by one and the same item of hardware. The mere fact that certain measures are recited in mutually different dependent claims does not indicate that a combination of these measures cannot be used to advantage.

Claims

1. An automatic gain control system for a sound reproduction apparatus comprising: - input means for receiving an audio signal comprising a speech signal and noise, said audio signal being in the form of audio input frames, means for receiving an audio input frame and data indicating that said audio input frame includes a speech signal, and means for determining a gain value representative of a gain to be applied to said audio input frame, wherein said gain value is updated relative to the gain value representative of the gain applied to a previous audio input frame to take into account the signal level of said input frame, if said audio input frame includes a speech signal, and, otherwise, said gain value is held constant at the gain value representative of the gain applied to said previous audio input frame.
2. A system according to claim 1, further comprising clipping means for clipping the audio signal at a predetermined clipping level.
3. A system according to claim 2, wherein said sound reproduction apparatus includes an input buffer having a maximum peak level, and the gain value is calculated as the clipping level divided by the maximum peak level of the input buffer.
4. A system according to claim 1, further comprising means for limiting the maximum resultant gain to be applied to the audio input frame to a predefined level.
5. A system according to claim 1, further comprising means for smoothing the gain value if the input audio frame includes a speech signal.
6. A method for performing automatic gain control in a sound reproduction apparatus having an input means for receiving an audio signal comprising a speech signal and noise, said audio signal being in the form of audio input frames, the method comprising the steps of: receiving an audio input frame and data indicating that said audio input frame includes a speech signal, determining a gain value representative of a gain to be applied to said audio input frame, wherein said gain value is updated relative to the gain value representative of the gain applied to a previous audio input frame to take into account the signal level of said audio input frame, if said audio input frame includes a speech signal, and otherwise, said gain value is held constant at the gain value representative of the gain applied to said previous audio input frame.
PCT/IB2006/053901 2005-10-26 2006-10-24 Adaptive volume control for a speech reproduction system WO2007049222A1 (en)

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CN104579212A (en) * 2015-01-30 2015-04-29 青岛海信电器股份有限公司 Method and device for adjusting audio gains
CN109716432A (en) * 2018-11-30 2019-05-03 深圳市汇顶科技股份有限公司 Gain process method and device thereof, electronic equipment, signal acquisition method and its system
US10701483B2 (en) 2017-01-03 2020-06-30 Dolby Laboratories Licensing Corporation Sound leveling in multi-channel sound capture system

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Cited By (11)

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WO2013162329A1 (en) * 2012-04-28 2013-10-31 Samsung Electronics Co., Ltd. Apparatus and method for outputting audio
CN104272599A (en) * 2012-04-28 2015-01-07 三星电子株式会社 Apparatus and method for outputting audio
US9641145B2 (en) 2012-04-28 2017-05-02 Samsung Electronics Co., Ltd. Apparatus and method for outputting audio according to audio tables
CN104272599B (en) * 2012-04-28 2017-06-20 三星电子株式会社 Apparatus and method for exporting audio
WO2014043024A1 (en) * 2012-09-17 2014-03-20 Dolby Laboratories Licensing Corporation Long term monitoring of transmission and voice activity patterns for regulating gain control
US9521263B2 (en) 2012-09-17 2016-12-13 Dolby Laboratories Licensing Corporation Long term monitoring of transmission and voice activity patterns for regulating gain control
CN104579212A (en) * 2015-01-30 2015-04-29 青岛海信电器股份有限公司 Method and device for adjusting audio gains
CN104579212B (en) * 2015-01-30 2017-04-19 青岛海信电器股份有限公司 Method and device for adjusting audio gains
US10701483B2 (en) 2017-01-03 2020-06-30 Dolby Laboratories Licensing Corporation Sound leveling in multi-channel sound capture system
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