WO2014209434A1 - Procédés et systèmes d'amélioration de la voix - Google Patents

Procédés et systèmes d'amélioration de la voix Download PDF

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Publication number
WO2014209434A1
WO2014209434A1 PCT/US2014/016980 US2014016980W WO2014209434A1 WO 2014209434 A1 WO2014209434 A1 WO 2014209434A1 US 2014016980 W US2014016980 W US 2014016980W WO 2014209434 A1 WO2014209434 A1 WO 2014209434A1
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WIPO (PCT)
Prior art keywords
audio
noise
sound
voice
preset
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PCT/US2014/016980
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English (en)
Inventor
Lloyd Trammell
Original Assignee
Max Sound Corporation
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Publication of WO2014209434A1 publication Critical patent/WO2014209434A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1785—Methods, e.g. algorithms; Devices
    • G10K11/17857—Geometric disposition, e.g. placement of microphones
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1787—General system configurations
    • G10K11/17873—General system configurations using a reference signal without an error signal, e.g. pure feedforward
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1787—General system configurations
    • G10K11/17885—General system configurations additionally using a desired external signal, e.g. pass-through audio such as music or speech
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/10—Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H04R1/1083—Reduction of ambient noise
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/03—Synergistic effects of band splitting and sub-band processing
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2460/00—Details of hearing devices, i.e. of ear- or headphones covered by H04R1/10 or H04R5/033 but not provided for in any of their subgroups, or of hearing aids covered by H04R25/00 but not provided for in any of its subgroups
    • H04R2460/01—Hearing devices using active noise cancellation
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R27/00—Public address systems

Definitions

  • Embodiments of the present, invention relate to U.S. Provisional Application Serial No. 61/765,634, filed February 15, 2013, entitled “PROCESS FOR IMPROVING AUDIO”, the contents of which are incorporated by reference herein and which is a basis for a claim of priority,
  • equalization is the process commonly used to alter the frequency response of an audio system using linear filters. Most hi-fi equipment uses relatively simple filters to make bass and treble adjustments. Graphic and parametric equalizers have much more flexibility in tailoring the frequency content of an audio signal.
  • An equalizer is the circuit or equipment used to achieve equalization. Since equalizers adjust the amplitude of audio signals at particular frequencies, they are, in other words, frequency-specific volume knobs. 1
  • Equalizers are used in recording studios, broadcast studios, and live sound reinforcement to correct the response of microphones, instrument pick-ups, loudspeakers, and hall acoustics. Equalization may also be used to eliminate unwanted sounds, make certain instruments or voices more prominent, enhance particular aspects of an instrument's tone, or combat feedback (howling) in a public address system. Equalizers are also used in music production to adjust the timbre of individual instruments by adjusting their frequency content and to fit individual instruments within the overall frequency spectrum of the mix. 2
  • Tone control is a type of equalization used to make specific pitches or "frequencies" in an audio signal softer or louder
  • a tone control circuit is an electronic circuit that consists of a network of filters which modify the signal before it is fed to speakers, headphones or recording devices by way of an amplifier. 4
  • API is a low requirement, but high quality method and system for improving an original input audio.
  • the core technology is not exposed. There are only places to connect and control the API available to be used or seen,
  • API audio technology A manufacturer can implement the API audio technology into its own devices with ease, The API would be compiled for whatever format is needed, including custom ones. Using the API will greatly increase both the dynamic and frequency ranges of the audio passed through it.
  • API is dynamic, not static, process for greater precision and quality than ever before available to these types of uses.
  • the inventive API of is a computer implemented method for improving audio, which includes processing audio with a combination of the wave adjustment tool that is the subject of co-pending U.S. Patent Application No. 14/183,292, filed on February 18, 2014, entitled WAVE ADJUSTMENT TOOL (WAT); and a collection of preset settings for adjusting specific harmonic content to a specific genres of music.
  • WAT WAVE ADJUSTMENT TOOL
  • the preset settings is selected from the following criteria a) different genres (styles or types) of music: b) auto-preset, and c) a generic preset.
  • the auto preset value is determined by the metadata which is commonly included in an audio file.
  • the selected preset will be "Generic.” This particular genre preset is designed for optimal sound across many genres, making it generic.
  • FIG, 1 is a signal flow chart showing the Wave Adjustment Tool module according to an embodiment of the present invention.
  • FIG. 2 is a signal flow chart showing API processing of PCM digital audio for use on a DSP chip.
  • Audio input 100 is stored in a buffer.
  • Each preset 230 will select a specific genre of music.
  • Another preset can be aiiio-preset that is selected by genre in metadata.
  • Yet another preset is a single generic preset that covers all music.
  • Preset 230 consists of the modules below, with explanation of same.
  • controls 220 are not exposed in the API 210, only a name which represents the settings for that particular setting (preset). Explanation of the functions are provided below.
  • EXPAND 110 is a 4 pole digital low pass filter with an envelope follower for dynamic offset (fixed envelope follower), This allows the output of the filter to be dynamically controlled so that the output level is equal to whatever the input is to this filter section. For e.g., if the level at the input is -6dB, then the output will match that, Moreover, whenever there is a change at the input, the same change will occur at the output regardless of either positive or negative amounts.
  • the frequency for this filter is, e.g., 20 to 20k hertz, which corresponds to a full range.
  • EXPAND 110 The purpose of EXPAND 110 is to "warm up" or provide a fuller sound as audio 100 passes through it, The original sound 100 passes through, and is added to the effected sound for its output. As the input amount 100 varies, so does the phase of this section. This applies to all filters used in this software application. Preferably all filters are of the Buttsrworth type.
  • SPACE 120 refers to the block of three modules identified by reference numerals 121. 122 and 123.
  • the first module SPACE 121 - which follows EXPAND 110 envelope follower, sets the final level of this module, This is the effected signal only, without the original.
  • SPACE ENV FOLLOWER 122 tracks .the input amount and forces the output level of this section to match.
  • SPACE FC 123 sets the center frequency of the 4 pole digital high pass filter used in this section. This filter also changes phase as does EXPAND 110.
  • SPACE blocks 120 are followed by the SPARKLE 130 blocks. Like SPACE 120, there are several components to SPARKLE.
  • SPARKLE HPFC 131 is a 2 pole high pass filter with a preboost which sets the lower frequency limit of this filter. Anything above this setting passes through the filter while anything below is discarded or stopped from passing.
  • SPARKLE TUBE THRESH 132 sets the lower level at which the tube simulator begins working. As the input increases, so does the amount of the tube sound. The tube sound adds harmonics, compression and a slight bit of distortion to the input audio 100. This amount increases slightly as the input level increases.
  • SPARKLE TUBE BOOST 133 sets the final level of the output of this module, This is the effected signal only, without the original.
  • the SUB BASS 140 module lakes the input signal and uses a low pass filter to set the , upper frequency limit to about 100Hz.
  • An octave divider occurs in the software that changes the input signal to lower by an octave (12 semi tones) and output to the only control in the interface, which is the level or the final amount. This is the effected signal only, without the original
  • All of the above modules 110 to 140 are directed into SUMMING MIXER 160 which combines the audio.
  • the levels going into the summing mixer 160 are controlled by the various outputs of the modules listed above. As they all combine with the original signal 100 fed through the DRY 150 module there is interaction in phase, time and frequencies that occur dynamically, These changes all combine to create a very pleasing audio experience for the Listener in the form of "enhanced" audio content. For example, a change in a single module can have a great affect on what happens in relation to the other modules final sound or the final harmonic output of the entire software application,
  • input audio 200 can also be processed by Wave Adjustment Tool (WAT) 240.
  • WAT 240 allows for low, mid. and hi tone controls and, as noted above, is subject to a separate patent application,
  • Bypass 250 allows user to turn API 210 on or off. After API 210, the audio continues in the audio path and can be stored in a device or output as audio 260. By using Bypass 250 user can compare the treated versus the untreated audio.
  • API 210 provides superior quality audio from compressed formats such as MP3, AAC, etc.
  • API 210 allows for selection of specific presets and fine-tuning of audio within three frequency ranges with the Wave Adjustment Tool 240.
  • a user GUI can be graphically built to the customer preferences, within the limits of the device the API 210 is installed in. Additional custom presets can be created and inserted into the API 210.
  • API 210 includes a single shared library compiled for the target platform and delivered as a binary.
  • the iibrary is written in C++ and delivered with headers that can be used in either C or C++,
  • API 210 is inserted in the audio signal 200 path immediately prior to the output,
  • the host provides a single interleaved buffer or two mono buffers for processing,
  • the processed buffer(s) is returned to the host for further processing.
  • Buffers can be provided in linear PCM format and can be 8, 16, 24, or 32 bits.
  • the API processor 210 converts the samples to floating point prior to processing. The samples are then converted back to the input format. The process supports any sample rate; however changing from one sample rate to another requires creating a new instance of the object,
  • Wave Adjustment Tool 240 equalization (low, medium, and high) is also available, Changes to equalization can be made at the beginning of processing a new buffer.
  • the WAT 240 affects specific frequency ranges. At this time there are three control ranges (low, mid, high) that operate in both negative and positive amounts. These amounts are determined by the program. la all three controls, the frequencies, the widths, and the amoants are changed as the sliders move up and down (positive or negative) directions.
  • Control module is shown y reference numeral 220.
  • the control choices available to the manufacturer/end user on an API configured device can be, for example, any or all of the following:
  • the API 210 is very basic for installation aiid/or use.
  • the host application creates an instance of the object, passing in a default buffer size and sample rate.
  • the host then simply calls the process routine as audio buffers become available.
  • the manufacturer/end user can expect much more harmonic content and greater dynamic range than without the APL Almost like lifting a blanket off of your speakers.
  • Another advantage is thai the manufacturer can use less efficient, and less costing, components to achieve much better sound.
  • Embodiments of the present invention relate to U.S. Provisional Application Serial No, 61/765,619 filed February 15, 2013, entitled “ACTIVE NOISE CANCELLATION", the contents of which are incorporated by reference herein and which is a basis for a claim of priority.
  • the present invention relates to a method and device for enhancing an audio source by reducing and eliminating background and other ambient noise in the sound wave, especially in enclosed cabins such as those found io airplanes, auios. ships, trains, homes and similar structures.
  • Listening environments typically inciude interference from surrounding audio sources such as people, devices, motions, etc., which prevent a listener from experiencing the best sound from the intended audio source.
  • the term noise cancellation or nois control is conventionally used to describe the process of minimizing or eliminating sound emissions from sources that interfere with the listeners' intended audio source, often for personal comfort, environmental considerations or legal compliance.
  • Conventional active noise reduction techniques involve recognizing the noise in the transmitted or received signal 5 .
  • the noise signal is recognized, it is reduced and removed by subtracting it from the transmitted or received signal.
  • This technique is implemented using a digital signs! processor (DSP) or software.
  • DSP digital signs! processor
  • Adaptive algorithms are designed to analyze the waveform of the background aural or non-aural noise, then based on the specific algorithm generate a signal that will either phase shift or invert the polarity of the original signal.
  • This inverted signal (in anti-phase) is then amplified and a transducer creates a sound wave directly proportional to the amplitude of the original waveform, creating destructive interference. This effectively reduces the volume of the perceivable noise 6
  • a Jioise-caiicellaiion speaker may be co-located with the sound source to be attenuated. In this case it must have the same audio power level as the source of the unwanted sound.
  • the transducer emitting the cancellation signal may be located at the location where sound attenuation is wanted (e.g. the user's ear). This requires a much lower power level for cancellation but is effective only for a single user/
  • the conventional noise reduction systems suffer from many deficiencies. For example, noise cancellation becomes more difficult as the three dimensional wave-fronts of the unwanted sound and the cancellation signal could match and create alternating zones of constructive and destructive interference, reducing noise in some spots while doubling noise in others.
  • small enclosed spaces e.g. the passenger compartment of an automobile
  • global noise reduction can be achieved via multiple speakers and feedback microphones; and measurement of the modal responses of the enclosure, but redaction in larger spaces is more problematic, 8
  • the Active Noise Cancellation (“ANC”) of the present invention is a system consisting of both analog and digital components that is specifically designed for reducing and eliminating ambient noise in an enclosed cabin environment of various sizes and shapes, such as those found in aircrafts, ships, trains, automobiles and even homes,
  • the method and system is dynamic in that it continuously monitors and changes as the ambient noise in the cabin changes,
  • the inventive ANC system includes two or more microphones that are placed in the target cabin in which noise reduction is sought, preferably the microphones are situated in equal distances in the horizontal and perpendicular directions corresponding to a two-dimensional plane. Each microphone monitors sound wa es in its corresponding zone and the overlaps of any of its surrounding zones. The number of microphones and zones will be determined by the size of the enclosed cabin the system is used in. Preferably, the microphones are of the Cardioids type.
  • the signals from the microphones are fed to an analog to digital converter, which converts the analog signals received from the microphones to digital signals.
  • the converted digital audio is analysed for content and ambient noise is identified for further processing.
  • the ambient noise is monitored fox changes. There could be a single or multiple noise frequencies that are identified and subsequently monitored,
  • Phase Modulator dynamically changes the phase of the ambient noise, always in a negative amount, of the digital audio received.
  • the negative phase sound is added back to the original noise which results in a reduction or cancellation of the sound wave corresponding to the noise.
  • FIG. 3(a) is a block diagram of an exemplary embodiment of the Active Noise Cancellation Module according to the present invention.
  • FIG, 3(b) is a block diagram of an exemplar)' embodiment showing a system incorporating Active Noise Cancellation Module according to the present invention.
  • FIG. 4 is an illustration of an exemplary application of the Active Noise Cancellation Module according to the present invention.
  • FIGS. 5(a) and 5(b) are exemplary illustrations of how the inventive process determines and differentiates noise from desirable audio
  • FIG. 3(a) - 3(b) An, embodiment of the operation of the Active Noise Cancellation technique of the present invention is depicted in the block diagram of Figures 3(a) - 3(b), Preferably, the inventive ANC process is performed by a single module identified by reference numeral 130 in the system shown in the block diagram of Figure s 3(a) - 3(b).
  • multiple microphones 100 provide the input audio source received for further analysis and processing
  • the microphones are of the Cardioid type.
  • the microphones are spaced in the enclosed cabin in which the noise reduction is being performed in equal distances from each other in a two dimensional lateral and perpendicular directions,
  • A/D converter 110 Analog-to-digital converter 110
  • the converted digital audio from the A D converter 110 is fed to the inventive Noise Cancelling Processor (CP) module 120 for processing.
  • the Noise Cancellation Processor Module 120 performs several steps on the sound wave it receives from the A D converter which will ultimately result in an audio sound with reduced or cancelled ambient noise levels.
  • A/D converted audio sound 110 is analyzed for content and ambient noise is identified. Once the noise wave is identified, it is further analyzed for frequency, amplitude and phase values.
  • the Compare step 132 monitors the amplitude, frequency and phase of the original sound wave for changes to ambient noise are subsequently performed as needed to identify any additions or changes to the determined noise.
  • the Change step 133 identifies any changes that are needed to be made to the incoming digital noise in both positive and negative direction, in the identified ambient noise.
  • Phase Modulator step 140 dynamically changes the phase of the identified ambient noise, in a negative amount, and creates a new noise correction wave based on the digital audio received. These changes are dynamic and self adjusting in nature,
  • Phase Modulator Audio Output step 150 is a phase modulated audio output (digital or analog) that feeds into the existing audio system in the enclosed cabin.
  • the modified noise output from the Phase Modulator 130 is added back to the original noise in a phase shift of 90 to 180 degrees as needed to cancei out the input noise.
  • the resulting combination of the original noise sound waves and the newly created noise correction wave will result in a reduction and cancellation of the noise present in the original audio sound.
  • This Phase Modulation is a constantly changing amount. The amount of change is derived from the analyzing of the input noise and its amplitude plus harmonic content.
  • FIG. 4 shows an exemplary embodiment of the. present invention in an airplane setting.
  • microphones 300 are placed equidistantly in the lateral and perpendicular directions. Zones 310 corresponding to microphones 300 are identified,
  • FIGS. 5(a) and 5(b) show an exemplary illustration of how the inventive process determines and differentiates noise from desirable audio.
  • the Figures show examples of audio that includes a small amount of noise, 310 and 320 refer to the desired audio in this example.
  • 330 identifies ihe audio noise in this example, which is also identified by the circles in Figure 5(b), This particular noise is about 15,5 kHz with a narrow bandwidth, as most noise is. This spike will continue to appear through the audio clip thus identifying ii as something that is constant and needs to be removed.
  • a Noise Cancellation Process for enclosed cabins is disclosed.
  • an input audio source corresponding to sound received from multiple microphones situated equidistantly in both directions in a two dimensional plane, is converted to a digital signal via an analog to digital (A/D) converter.
  • the A'T> converted audio is analyzed for content to identify ambient noise.
  • the frequency, amplitude and phase of the idendfted ambient noise is subsequently determined.
  • a Noise correction sound wave is generated with negative phase of that corresponding to the identified ambient noise, The noise correction sound wave is added to the identified noise to create a noise corrected sound.
  • Embodiments of the present invention relate to U.S. Provisional Application Serial No. 61/765,637, filed February 15, 2013, entitled “VOICE CALL .ENHANCEMENT”, the contents of which are incorporated by reference herein and which is a basis for a claim of priority,
  • Sound quality is typically an assessment of the accuracy, enjoyability, or clarity of audio output from an electronic device. Quality can be measured objectively, such as when tools are used to measure a certain aspect of quality with which the device reproduces an original sound; or it can be measured subjectively, such as when human listeners respond to the sound or gauge its perceived similarity to another sound. 9
  • [ ⁇ ] Tbe sound quality of a reproduction or recording depends on a number of factors, including the equipment used to make it, processing and mastering done to the recording, th equipment used to reproduce it, as well as the listening environment used to reproduce it. in some cases, processing such as equalization, dynamic range compression or stereo processing may be applied to a recording to create audio thai is significantly different from the original but may be perceived as more agreeable to a listener. In other cases, the goal may be to reproduce audio as closely as possible to the original 10
  • audio quality may refer to proper placement of microphones around a room to optimally use room acoustics
  • Human voice has a frequency range thai extends from 80 Hz to 14 kHz.
  • traditional, voice band or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz.
  • humaars communicate over telephone lines, there is resulting loss of quality in the voice heard through phone lines due to the loss in the frequency range,
  • a computer implemented method for enhancing processed voice includes receiving voice audio and enhancing the voice audio in multiple harmonic and dynamic ranges.
  • the audio is enhanced by resynthesizing the audio into full range PCM wave.
  • the received voice audio can be in compressed format.
  • the voice audio can be, for example, from an inbound phone call or from an outbound phone call
  • FIG. 6 is a block diagram of an exemplary embodiment of the Voice Call Enhancement process of the present invention corresponding to an inbound and an outbound call,
  • FIG. 7 is a block diagram showing the various processing steps of an embodiment of the present invention.
  • FIG. 8 is an example of the settings corresponding to various processing steps of the present invention for an android application.
  • the inventive voice enhancement process is used to help clarify both inbound and outbound voice calls on telephonic communication devices. This goal is accomplished by restoring (resynthesizirtg) the audio to a much greater harmonic and dynamic range than the original audio.
  • the voice enhancement module 120 resynthesizes the harmonic and dynamic properties of the received audio into a full range PCM (Pulse-code modulation) wave with extended audio content. The result is added clarity to the compressed, band limited audio of the incoming audio.
  • the enhanced voice signal is then received by the phone speaker 130 and transmitted to user 140.
  • the transmitted wave retains much of the quality of the original voice, even after being compressed by the cell phone system.
  • the initial audio signal 200 is subjected to parallel processing by four module processors identified as EXPAND 210, SPACE 220, SPARKLE 230 and SUB PASS 240, and is then combined with the original audio source in a mixer 250.
  • the unprocessed original audio 200 is received by a selector DRY 250, which sets the amount of the original audio source 200 in the mixer.
  • DRY 250 can have a preset control, such as in the range of about 0 tol, in 0.1 increments.
  • EXPAND 210 is a 4 pole digital low pass filter with an envelope follower for dynamic offset (fixed envelope follower). This allows the output of the filter 210 to be dynamically controlled so that the output level is equal to the input to this filter section. For example, if the level at the input is -6dB, then the output will match that amount. Moreover, changes at the input level result in the same change to occur at the output in either positive or negative amounts.
  • the frequency for this filter 210 is 40K to 20k hertz, which corresponds to a full range. In one embodiment, the frequency is about 2000 Hertz.
  • the range for EXPAND 210 is 0 tol, in intervals of 0.1.
  • EXPAND 210 is preset in the program.
  • the purpose of this filter 210 is to "warm up” or provide a fuller sound as audio that passes through it. The original sound passes through, and is added to the effected sound for its output. As the input amount increases or decreases (varies), so does the phase of this section. This applies to all filters used in this software application, which, preferably are of the Buttenvorth type.
  • SPACE 220 is an envelope controlled bandpass filter and includes three sub processing steps.
  • SPACE 221 corresponds to the output level for this block
  • SPACE ENV FOLLOWER 222 is the envelope follower modulation amount.
  • SPACE FC 222 corresponds to the frequency range for SPACE 220 block, in one embodiment, the output amplitude for SPACE 220 is between about 0 to 3, preferably about 1,8 and the frequency range for SPACE 220 is between about 1000 to about 8000 Hertz,
  • the settings for SPACE can also be preset,
  • SPACE 220 there are several components to SPACE 220.
  • SPACE 221 is the amount is after the envelope follower and sets the final level of this module. This is the processed signal only, without the original.
  • SPACE ENV FOLLOWER 222 tracks the input amount and forces the output level of this section to match.
  • SPACE FC 223 sets the center frequency of the 4 pole digital high pass filter used in this section. This filter also changes phase as does EXPAND 210.
  • the original audio signal 200 is also processed by SPARKLE. 230, which is a high pass filter.
  • FIG. 7 depicts three blocks corresponding to SPARKLE 230.
  • SPARKLE HPFC 231 is the output level for this block which sets HP filter frequency.
  • SPARKLE TUBE THRESHOLD 232 sets the threshold frequency amount of tube simulator sound,
  • the frequency for the high pass filter can be about 4000 to about 10000 Hertz,
  • the tube simulator can be set in single digits from 1-5,
  • the threshold can range from 0-1 in 0.1 intervals.
  • the settings for SPARKLE 230 can also be preset,
  • SPARKLE 230 includes three sub processing steps.
  • SPARKLE HPFC 231 is the output level for this block, which sets HP filter frequency.
  • SPARKLE TUBE THRESH 232 sets the lower level at which the tube simulator begins working. As the input increases, so does the amount of the tube sound. The tube sound is adding harmonics, compression and a slight bit of distortion to the input sound. This amount increases slightly as the input level increases.
  • SPARKLE TUBE BOOST 233 sets amount of tube simulator sound, In one embodiment, the frequency for the high pass filter can be about. 4000 to about 10000 Hertz.
  • the tube simulator can be set in single digits from 1-5.
  • the threshold can range from 0 tol in 0,1 intervals.
  • the settings for SPARKLE can also he preset,
  • the original audio signal 200 is also processed by SUB BASS 240, which operates to add an amount of dynamic synthesized sub bass to the audio.
  • the frequency of the subpass is about 120Hz to less.
  • SUB BASS 240 operates on the input signal 200 and uses a low pass filter to set the upper frequency limit to about 100Hz.
  • An octave divider occurs in the software that changes the input signal to lower by an octave (12 semi tones) and output to the only control in the interface, which is the level or the final amount. This is the effected signal only, without the original.
  • Processed audio from the above modules are fed into a summing mixer 250 which combines the audios,
  • the levels going into the summing mixer are controlled by the various outputs of the modules lisi&c above. As they all combine with the unprocessed original signal 260, there is interaction in phase, time and frequencies that occur dynamically. These changes all combine to create a very pleasing audio experience for the listener in the form of "enhanced" audio content. For example, a change in a single module can have a great affect on what happens In relation to the other modules finaj sound or the final harmonic output of the entire software application.
  • This process can be a small program or API for use m any smart phone format for fixed processor or for floating point processors, or used in any device that needs voice enhancement or clarity,
  • FIG. 8 is a table that illustrates an example of a setting for an android application according to an embodiment of the present invention.
  • the table shows various settings corresponding to some of the processing modules of FIG. 7 for Voice and Music 310, Female Voice 320 ⁇ Male Voice 330 and Male and Female 340,
  • a Voice Call Enhancement Method for wireless telephonic comni nicaiioii devices includes providing an input voice audio source, enhancing the voice audio input in multiple harmonic and d namic ranges and outpuUing the voice enhanced audio.
  • the Voice Call Enhancement method is suitable for use of wireless telephony devices, such as cellular phones.
  • the enhancement includes resynihesizirsg audio to as increased harmonic and dynamic range than original valises,
  • Embodiments of the present invention relate to U.S. (Provisional/CIP .%) Application Serial No. 61/765,620, filed February 15, 2013, entitled “VOICE RECOGNITION ENHANCEMENT", the contents of which are incorporated by reference herein and which is a basis for a claim of priority,
  • Human voice has a frequency range that extends from 80 Hz to 14 kHz.
  • traditional, voice band or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz.
  • audio frequencies limit audio frequencies to the range of 300 Hz to 3.4 kHz.
  • Wideband audio also known as HD voice, refers to the "next generation" of voice quality for telephony audio resulting in high definition voice quality compared to standard digital telephony "toll quality”.
  • HD voice extends the frequency range of audio signals transmitted over telephone lines, resulting in an expanded frequency range and therefore higher quality speech.
  • Typical wideband audio systems relax the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz or higher.
  • communication devices such as cellular phones, which rely on limited narrow band widths, have transmission that is very limited in its audio range. Due to this limitation in the available frequency range, manufacturers of telephonic communication devices will only make devices that operate within this criteria. As an example, ceil phone manufacturers would not manufacture a full 20 to 20kHz audio capable phone, as it would not cost efficient since the improvement could not be above what the transmission is capable of. At this time, wideband is not yet a commonly used format. [00088] Due to the limited range of available bandwidth, teleconununication devices that rely on such bandwidth, such as cell phones, utilize electronics and circuitry thai have a very narrow frequency range. This limited range results in anything from degraded to garbled voice quality on the receiving user.
  • DSP digital signal processing
  • Voice intelligibility is, among other factors, dependent upon consonant recognition. Most consonants have percussive leading edges. So, for example, by enhancing these consonants, the process makes speech more intelligible. Moreover, the level of such increase would be small which will prevent an increase in reverberation, as, for example, would be the case with simple equalization. TSie effect helps intelligibility in a noisy environment as well by supplying more cues. The benefits are realizable from full response systems to low fidelity telephones. Tuning, of course, would be different for different applications.
  • the inventive Voice Recognition Enhancement includes a harmonics generator that looks' for transients in the input voice signal and generates more harmonics on those transients, essentially enhancing the transients while leaving the non-transient material untouched.
  • the VRE improves the "source” that feeds the specific telephony product thereby allowing the product to perform as the manufacture intended and is not limited due to compressed sound files.
  • FIG, 9 is a block diagram of an exemplary embodiment of the Voice Recognition Enhancement method of the present invention corresponding to an inbound telephone call.
  • FIG, 10 is a block diagram of an exemplary embodiment of the Voice Recognition Enhancement method of the present invention corresponding to an outbound telephone call
  • FIG.l l is a depiction of signals corresponding to a typical voice call from a cell phone.
  • FIG. 12 is a depiction of signals corresponding to a typical voice call from a cell phone that has been processed by the Voice Recognition Enhancement method of the present invention.
  • FIG. 9 An embodiment of the operation of the Voice Recognition Enhancement Method asd system of the present invention is depicted in the block diagram of Figure 9,
  • the inventive VRE process is performed by a single processor module identified by reference numeral 120 in the system shown in the block diagram of Figure 9 corresponding to an incoming call, and reference numeral 210 in the outbound set up shown m Figure 10.
  • inbound call 100 is received by a telephony through a microphone 110.
  • Signal from the microphone 110 is fed to the inventive VRE processor, where the sound signal is processed for enhancement.
  • Voice enhancement at this step is accomplished by restoring (resynthesizing) the inbound voice audio to a much greater harmonic and dynamic range than that possessed by the original voice signal. For example, an incoming voice signal with a 16 bit audio range can be expanded into a 20 bit range.
  • utilizing this process requires no change in the hardware of the receiving device.
  • the harmonic and dynamic properties of the voice signal are resynihesized into a full range PCM (Pulse-code modulation) wave with extended audio content. More harmonic and dynamic information is generated resulting in extended (increased) audio content. This, in turn, provides much more clarity to the compressed, band limited audio available in the existing cell audio.
  • PCM Pulse-code modulation
  • Figure 10 shows a corresponding exemplary application of the inventive VRE process for an outbound call.
  • user speaks into the device's microphone for an outbound call 200
  • Sound waves corresponding to the voice of the caller are subsequently fed to and are processed by the inventive VRE module 210, where they are enhanced as described above prior to being sent out of the device to a call receiver 220.
  • the resulting VRE processed sound is much clearer, more real sounding wave that is transmitted to the call receiver.
  • the transmitted wave retains much of the quality of the original voice, even though it has io be compressed by the cell phone system,
  • the Voice Enhancement Process of the present invention can be used with any conventional voice recognition system, including those not associated with making phone calls. These include for example voice dictation and use of programs that respond to voice (such as SIRI).
  • Figures 11 and 12 correspond to images of a sound waves 300 and 310, corresponding to a voice call from a cellular phone prior to and following processing by the inventive VRE process.
  • Reference numeral 300 corresponds to the pre-processed sound
  • refereiice numeral 310 corresponds to the sound 300 that has been processed by the inventive. From the two graphic examples of a voice call without and with the Voice Call Enhancement it is clear that material has been resynthesized into the processed wave, thus making it much clearer and much more discernible to the listener. In the provided examples, from left to right represents frequency range 0 Hz to 20 kHz and amplitude range of -140 to 0 DBFS. The FFT size is 8192 and the FFT type is Biackrnan - Harris.
  • a Voice Recognition Enhancement Method for wireless telephonic communication devices includes providing an input voice audio source-, enhancing the voice audio input in one or more of harmonic arsd dynamic ranges and ouip3 ⁇ 4iting the voice enhanced audio.
  • the Voice Recognition Enhancement method is suitable for use of wireless telephony devices, such as cellular phones.
  • the enhancement includes re-synthesizing audio to an increased harmonic and dynamic range ftars original values.
  • Embodiments of the present invention relate to U.S. Provisional Application Serial No. 61/765,631, filed February 15, 2013, entitled “PERSONAL ACTIVE NOISE CANCELLATION", the contents of which are incorporated by reference herein and which is a basis for a claim of priority.
  • the present invention relates to a personal, portable, sound control system which reduces background noise levels and provides a quieter sound environment for the user for work or relaxation.
  • the inventive system and method is suitable for use at any location where user utilizes for work or relaxation, such as office cubicles, hotel rooms, office space at home, and the like.
  • Personal environments typically utilized for work or relaxation often include interference from surroundi g audio sources such as people, devices, motions, etc., which interfere with the person's work or relaxation.
  • surroundi g audio sources such as people, devices, motions, etc.
  • hotel guests occupying rooms adjacent to roads are often prevented from getting a restful night by or distracted from work or relaxation during the da by the sounds of moving vehicles using the road,
  • workers using cubicles in office environments are often distracted by noise from nearby offices and cubicles.
  • noise cancellation or noise control is used to describe the process of minimizing or eliminating sound emissions from sources that interfere, with the listeners' quiet or intended audio source, often for personal comfort, environmental considerations or Segal compliance.
  • Sound is a pressure wave, which consists of a compression phase and a rarefaction phase.
  • a noise-cancellation speaker emits a sound wave with the same amplitude but with inverted phase (also known as antiphase) to the original sound.
  • the waves combine to form a
  • a noise-cancellation speaker may be co-located with the sound source to be. attenuated. In this case it must have the same audio power level as the source of the unwanted sound.
  • the transducer emitting the cancellation signal may be located at the location where sound attenuation is wanted (e.g. the user's ear). This requires a much lower power level for cancellation but is effective only for a single user,' 5
  • Noise cancellation at other locations is more difficult as the three dimensional wavefronts of the unwanted sound and the cancellation signal could match and create alternating zones of constructive and destructive interference, reducing noise in some spots while doubling noise in others.
  • small enclosed spaces e.g. the passenger compartment of a car
  • global noise reduction can be achieved via multiple speakers and feedback microphones, and measurement of the modal responses of the enclosure/ 6
  • Applications can be "1 -dimensional" of 3-dimensional, depending on the type of zone to protect. Periodic sounds, even complex ones, are easier to cancel than random sounds due to the repetition in the wave form. 57
  • the inventive Personal Noise Reduction (PNR) method and system of the present application is specifically designed for reducing and eliminating ambient noise in a smali personal environment such as an office cubicle, a room in a home, a hotel room and the like,
  • the inventive PNR system includes two or more microphones that are placed in the target cabin in which noise reduction is sought, preferably the microphones are situated in equal distances as needed in a one dimensional arrangement or, in the horizoniai and perpendicular directions corresponding to a two -dimensional plane.
  • the number of microphones is determined by the size of the space the system is used in.
  • the microphones are of the Cardioids type.
  • Signals from the microphones are fed to an analog to digital converter, which converts the analog signals received from the microphones to digital signals, The converted digital audio is analyzed for content and ambient noise is identified for further processing. Signals from the microphones are also monitored for changes to the ambient noise. There could be a single or multiple noise frequencies that are identified and subsequently monitored,
  • DSP dynamically changes the phase of the ambient noise, always in a negative amount, of the digital audio received, The negative phase sound is added back to the original noise which results in a reduction or cancellation of the sound wave corresponding to the noise.
  • FIG. 13 is a schematic diagram of an exemplary embodiment of the arrangement of speakers and microphones in the Personal Noise Reduction system of the present invention.
  • FIG. 14 is a block diagram of an exemplary embodiment showing a system incorporating Personal Noise Reduction Module according to the present invention.
  • FIG. 13 An embodiment of the operation of the Personal Noise Reduction method of the present invention is depicted in the block diagram of Figure 13.
  • the inventive PNR process is performed by a single DSP processor module identified by reference numeral 210 in the system shown in the block diagram of Figure 14,
  • multiple microphones 100 receive sound from the environment and provide the input audio source for farther analysis and processing.
  • the microphones are of the. Cardioid type.
  • the microphones are two or more in number and are spaced in the persona! space targeted for noise reduction in equal distances from each other in a one or two dimensional arrangement as needed.
  • A/'D analog-to-digital converter
  • the converted digital audio from the A/D converter is fed to a digital sound processing module (referenced as DSP in FIG. 14) 210 for processing,
  • DSP digital sound processing module
  • the Personal Noise Reduction Module processes the received sound wave
  • This process will "listen" to the audio in an environment within a limited range (band pass). Any content in this area will be measured, or sampled, for amplitude. Preferably, the sound measurements wiii be made at about every 60 seconds.
  • phase inversion will be fixed to a constant 180 degrees
  • the inventive Personal Noise Reduction method has many applications. For example. When a guest is staying the night at a hotel close to an airport, By using this system user could remove a substantial amount of the exterior noise that enters his/her room, thus having a quieter room in the area where this device is used, It is intended for single room spaces only. Likewise, if a user is in an office cubicle with noisy surrounding, the inventive PN r R system will reduce surrounding noise and provide a much more quiet environment for work.
  • a Personal Noise Reduction for enclosed cabins is disclosed.
  • an input audio source corresponding to sound received from muiiiple microphones situated equidistantly in both directions in a two dimensional plane, is converted to a digital signal via an analog to digital (A D) convenor.
  • the AJO converted audio is analyzed for content to identify ambient noise, The frequency, amplitude and phase of the identified ambient noise is subsequently determined, A Noise, correction sound wave is generated with negative phase of that corresponding to the identified ambient noise.
  • the noise correction sound wave is added to the identified noise to create a noise corrected sound.
  • Embodiments of the present invention relate to U.S. Provisional Application Serial No. 61/766,077, filed February 18, 2013, entitled “WAVE ADJUSTMENT TOOL (WAT)", the contents of which are incorporated by reference herein and which is a basis for a claim of priority.
  • equalization is the process commonly used to alter the frequency response of an audio system using linear filters. Most hi-fi equipment uses relatively simple filters to make bass and treble adjustments, Graphic and parametric equalizers have much more flexibility in tailoring the frequency content of an audio signal
  • An equalizer is the circuit or equipment used to achieve equalization. Since equalizers, adjust the amplitude of audio signals at particular frequencies, they are, in other words, frequency-specific volume knobs. 21
  • Equalizers are used in recording studios, broadcast studios, and live sound reinforcement to correct the response of microphones, instrument pick-ups, loudspeakers, and hall acoustics. Equalization may also be used to eliminate unwanted sounds, make certain instruments or voices more prominent, enhance particular aspects of an instrument's tone, or combat feedback (howling) in a public address system. Equalizers are also used in music production to adjust the timbre of individual instruments by adjusting their frequency content and to fit individual instruments within the overall frequency spectrum of the mix. 2 "
  • Tone control is a type of equalization used to make specific, pitches or "frequencies in an audio signal softer or loader
  • a tone control circuit is an electronic circuit that consists of a network of filters which modify the signal before it is fed to speakers, headphones or recording devices by way of an amplifier.
  • Conventional tone control method is thus a static setting that can increase or decrease a fixed amount at a single frequency and bandwidth, While this does allow the user to customize a sound to his preference, as soon as anything changes this setting may not be desirable and the user will either accept compromise or be continually changing the amounts as different content is played, 2 *
  • a Wave Adjustment Tool (WAT) process and system for customizing sound is provided.
  • an input audio sound is received and is treated by a tone adjusting circuit that comprises three sections, including a first section for adjusting a low frequency tone; a second section for adjusting a mid frequency tone; and a third section for adjusting a high frequency tone, Audio processed by the first, second and third sections are mixed to produce an output audio sound.
  • the low frequency tone has a frequency of 100 Hz and a bandwidth of 0,5,; the mid frequency tone has a frequency of Z500 Hz and an adjustable bandwidth; and the high frequency tone has a frequency of 10 KHz and an adjustable bandwidth.
  • the inventive Wave Adjustment Tool is a different approach in that it dynamically monitors the audio content and adjusts itself to compensate for these changes in both a positive and negative direction. The end result is very pleasing and a more natural sound of the content being played.
  • the WAT process is not limited only three bands, More dynamic bands may be added as desired by programming them into the process and assigning the frequency, band width, and amount of dynamic change to be allowed per band. In this case it is a digital process, but it may be hardware (analog) if desired in any output format (mono, stereo, 5.1, 7, 1. etc)
  • FIG. 15 is a block diagram of an exemplary embodiment of the Wave Adjustment Tool according to the present invention.
  • FIG. 16 shows a typical use/implementation of the inventive Wave Adjustment Tool according to the present invention.
  • the Wave Adjustment Tool of the present invention is an improvement over the conventional tone adjustments in part because it is based on a dynamic approach that monitors the content of the received audio and adjusts itself to compensate for any changes in both a positive and negative direction. The end result is very pleasing and a more natural sound of the content being played.
  • the WAT ⁇ is not limited only three bands. More dynamic bands may be added as desired by programming them into the process and assigning the frequency, band width, and amount of dynamic change to be allowed per band. In this case it is a digital process, but it may be hardware (analog) if desired in any output format (mono, stereo, 5.1, 7, 1, etc.)
  • WAT is a three section tone adjusting circuit with some dynamic control.
  • the sections are LOW (bass), MID, and HIGH (treble).
  • input audio 100 is received for processing by the Wave Adjustment Tool of the present invention.
  • Input audio 100 is processed in parallel by the three sections of the WAT tone adjusting circuit, which include the LOW 110, MID 120 and HIGH 130 sections, The audio processed by the three sections (shown by reference numerals 140, 150 and 160 in FIG, 15) are then mixed to form output audio 170.
  • the LOW section has a frequency of 100Hz and a 0.5 bandwidth; MID has a frequency of 2500Hz with an adjustable bandwidth; and HIGH has a 10 kHz frequency and a ,5 bandwidth.
  • the center frequency is dynamically moved in both positive and negative amounts according to the input level of this bandpass filter.
  • the range is from 1.7 kHz on the low end to 4.5 kHz on the upper end with 2.5 kHz as the center or nominal setting,
  • the bandwidth will change.
  • the bandwidth will increase, for e.g., to a .5, while a positive change will decrease, for e.g., to a .1. This provides a larger frequency change for negative and a smaller, more precise change for positive level amounts in the filtered audio content.
  • the center frequency is fixed, e.g., at 10 kHz, but the bandwidth changes dynamically in positive amounts as the input level changes. For negative amounts the bandwidth stays at, e.g., .5, when the level decreases the bandwidth goes only to a max bandwidth of e.g., .3,
  • FIG. 16 shows a typical use/implementation of the inventive WAT process for smartphone 200 and its graphic configuration or look.
  • the "WAT Zero” box appears indicating that a change has been made from the zero position
  • Pressing the WAT Zero button will return all of the faders to a zero position and the button will disappear until a change is made.
  • the Low, Mid and High adjustment bars are respectively identified by reference numerals 210, 220 and 230.
  • a Wave Adjustment Tool process and system for customizing sound is provided.
  • An input audio sound is received.
  • a tone adjusting circitit that comprises three sections, iriduding a first section for adjusting a low frequency tone; a second section for adjusting a mid frequency tone; and a third section for adjusting a high frequency tone. Audio processed by the first, second and third sections are mixed to produce an output audio sound.
  • the Sow frequency tone has a frequency of 100 Hz and a bandwidth of 0.5.
  • the mid frequency tone has a frequency of 2500 Hz and an adjustable bandwidth
  • the high frequency tone has a frequency of 10 KHz and an adjustable bandwidth.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Otolaryngology (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)

Abstract

La présente invention a trait à un procédé mis en œuvre par ordinateur permettant d'améliorer le son, et comprenant un traitement du son au moyen d'une combinaison d'un outil d'ajustement des ondes (WAT) et d'une série de paramètres préréglés pour ajuster un résidu harmonique spécifique à des genres de musique spécifiques. Le préréglage est sélectionné dans le groupe comprenant différents genres (styles ou types) de musique, un préréglage automatique et un préréglage générique. La valeur du préréglage automatique est déterminée par les métadonnées communément incluses dans un fichier audio. S'il n'y a pas de correspondance entre les noms de genres préréglés et celui qui se trouve dans les métadonnées, le préréglage sélectionné est « générique ». Ce préréglage de genre particulier est conçu pour un son optimal dans de nombreux genres, ce qui le rend générique.
PCT/US2014/016980 2013-02-15 2014-02-18 Procédés et systèmes d'amélioration de la voix WO2014209434A1 (fr)

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US201361765631P 2013-02-15 2013-02-15
US201361765620P 2013-02-15 2013-02-15
US61/765,619 2013-02-15
US61/765,631 2013-02-15
US61/765,620 2013-02-15
US671765637 2013-02-15
US61/765,634 2013-02-15
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