公开(公告)号：US20160266870A1

公开(公告)日：2016-09-15

申请号：US15165284

申请日：2016-05-26

Applicant: Tencent Technology (Shenzhen) Company Limited

Inventor： Xiayu Wu ,  Rong Wan

IPC: G06F3/16 ,  H03G7/00

CPC classification number: G06F3/165 ,  A63F13/54 ,  A63F13/814 ,  A63F2300/6063 ,  G10H1/46 ,  G10H2210/026 ,  G10H2250/371 ,  H03G7/002

Abstract: A sound effect adjusting method and apparatus are provided. The method includes: determining an initial sound effect intensity value corresponding to each object currently selected; determining, according to the initial sound effect intensity value corresponding to each object currently selected, a sound effect intensity value corresponding to each type of object currently selected; acquiring a state of each type of object currently selected; adjusting, according to the state of each type of object currently selected, the sound effect intensity value corresponding to each type of object; and combining, according to adjusted sound effect intensity values of all types of objects currently selected, sound effects corresponding to all the types of objects, and using the combined sound effects as a sound effect adjustment result.

Abstract translation: 提供了一种声音效果调节方法和装置。 该方法包括：确定对应于当前选择的每个对象的初始声音效果强度值; 根据与当前选择的每个对象相对应的初始声音效果强度值，确定与当前选择的每种对象对应的声音效果强度值; 获取当前选择的每种类型的对象的状态; 根据当前选择的每种类型的对象的状态调整对应于每种对象的声音效果强度值; 并根据当前选择的所有类型的对象的调整的声音效果强度值进行组合，对应于所有类型的对象的声音效果，并且使用组合的声音效果作为声音效果调整结果。

公开(公告)号：US20160261953A1

公开(公告)日：2016-09-08

申请号：US15157135

申请日：2016-05-17

Applicant: Caavo Inc

Inventor： Ashish Dharmpal Aggarwal ,  Sharath Hariharpur Satheesh

IPC: H04R3/04 ,  H03H17/04 ,  H04R1/08 ,  G10L21/0232 ,  H04R29/00 ,  H03G7/00

CPC classification number: H04R3/04 ,  G06F3/165 ,  G10L21/0232 ,  G10L25/51 ,  H03G3/20 ,  H03G3/3005 ,  H03G3/32 ,  H03G5/005 ,  H03G5/165 ,  H03G7/002 ,  H03H17/04 ,  H03H2017/0472 ,  H04R1/08 ,  H04R3/00 ,  H04R25/50 ,  H04R25/505 ,  H04R29/00 ,  H04R29/001 ,  H04R29/004 ,  H04R2430/01 ,  H04R2499/11

Abstract: Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

公开(公告)号：US09431985B2

公开(公告)日：2016-08-30

申请号：US14086980

申请日：2013-11-22

Applicant: APPLE INC.

Inventor： Christopher Sanders

IPC: H03G3/00 ,  H03G7/00 ,  H03G1/02 ,  G06F3/16

CPC classification number: H03G7/00 ,  G06F3/165 ,  H03G1/02

Abstract: Techniques for controlling the volumes of multiple audio output devices using a collective (master) volume control and an individual volume control for each audio output device. In one set of embodiments, each individual volume control can be configured to indicate the current absolute volume level of its corresponding audio output device. When the master volume control is manually adjusted, the individual volume controls can be automatically adjusted in a manner proportional to the manual adjustment of the master volume control. In addition, when an individual volume control is manually adjusted to a setting or value that exceeds the master volume control, the master volume control can be automatically adjusted to be equal to, or greater than, the manually adjusted setting for the individual volume control. In this scenario, the other individual volume controls can remain unchanged.

公开(公告)号：US09397629B2

公开(公告)日：2016-07-19

申请号：US14059669

申请日：2013-10-22

Applicant: Bongiovi Acoustics LLC

Inventor： Anthony Bongiovi ,  Philip Fuller ,  Glenn Zelniker

IPC: H03G5/00 ,  G06F3/16 ,  H03G3/00 ,  H03G5/16 ,  H03G9/00 ,  H03G9/02 ,  H04R3/04 ,  H03G5/02 ,  H03G7/00

CPC classification number: H03G3/002 ,  G06F3/165 ,  G10L21/02 ,  H03G5/025 ,  H03G5/165 ,  H03G7/007 ,  H03G9/005 ,  H03G9/025 ,  H04R3/04

Abstract: The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

公开(公告)号：US20160197590A1

公开(公告)日：2016-07-07

申请号：US14916509

申请日：2014-09-08

Applicant: DOLBY LABORATORIES LICENSING CORPORATION ,  DOLBY INTERNATIONAL AB

Inventor： Jeroen KOPPENS ,  Jeffrey RIEDMILLER ,  Kristofer KJOERLING ,  Alexander GROESCHEL ,  Holger HOERICH ,  Alan J. SEEFELDT

IPC: H03G3/30 ,  H03G7/00 ,  H03G9/12 ,  G10L19/16

CPC classification number: G06F3/165 ,  G10L19/167 ,  G10L21/0324 ,  H03G3/3005 ,  H03G3/3089 ,  H03G7/002 ,  H03G7/007 ,  H03G9/005 ,  H03G9/12 ,  H04S1/00 ,  H04S7/30 ,  H04S2400/13

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

公开(公告)号：US20160164480A1

公开(公告)日：2016-06-09

申请号：US15001311

申请日：2016-01-20

Applicant: Timothy Shuttleworth

Inventor： Timothy Shuttleworth

IPC: H03G7/00 ,  H03G11/04 ,  H03G9/00 ,  H04R29/00 ,  H03G9/02

CPC classification number: H03G7/002 ,  H03G7/007 ,  H03G9/005 ,  H03G9/025 ,  H03G11/04 ,  H04R29/001

Abstract: A method, apparatus, and system for measuring and analyzing the effects of dynamics modifying processors on a signal. This new approach utilizes statistical analysis techniques to provide a direct comparison and evaluation between the processed signal and the unprocessed signal's dynamic characteristics. The method identifies and quantifies Effective Dynamic Range, Clip Tolerance, Lower Limit Tolerance, Crest Factor, and Diminuendo Factor, using either peak or r.m.s values. In an alternative embodiment, the invention allows for user adjustment and control of the relative relationship of Crest Factor and Diminuendo Factor, which the user may perceive as loudness.

公开(公告)号：US09325286B1

公开(公告)日：2016-04-26

申请号：US13858818

申请日：2013-04-08

Applicant: Rawles LLC

Inventor： Jun Yang

IPC: H03G3/20 ,  H03G7/00

CPC classification number: H03G3/20 ,  H03G7/00 ,  H03G7/002 ,  H03G7/007

Abstract: Audio clipping is prevented by attenuating an audio signal in accordance with values retrieved from a gain table. Corresponding amplitude values of a stereo audio signal are evaluated to determine a maximum of the values. The amount by which the maximum exceeds a predetermined threshold is used to calculate a table index, which is used to retrieve a gain value from the gain table. The gain value is then applied to the audio signal. The gain table is configured so that increasing index values produce decreasing gain values.

Abstract translation: 通过根据从增益表检索的值衰减音频信号来防止音频剪辑。 对立体声音频信号的对应幅度值进行评估，以确定最大值。 使用最大值超过预定阈值的量来计算用于从增益表中检索增益值的表索引。 然后将增益值应用于音频信号。 增益表被配置为使得增加的索引值产生减小的增益值。

公开(公告)号：US09325285B2

公开(公告)日：2016-04-26

申请号：US14174562

申请日：2014-02-06

Applicant: Oticon A/S ,  Sennheiser Communications A/S

Inventor： Svend Feldt ,  Thomas Kaulberg ,  Torben Christiansen ,  Claus Benjaminsen

IPC: H03G5/00 ,  H03G3/00 ,  H03F1/32 ,  H04R3/04 ,  H03G7/00 ,  H04R27/00 ,  H04R3/00

CPC classification number: H03G3/00 ,  H03F1/327 ,  H03G7/002 ,  H04R3/005 ,  H04R3/04 ,  H04R27/00 ,  H04R2410/07

Abstract: An audio processing device comprises a multitude of electric input signals, each electric input signal being provided in a digitized form, and a control unit receiving said digitized electric input signals and providing a resulting enhanced signal. The control unit is configured to determine the resulting enhanced signal from said digitized electric input signals, or signals derived therefrom, according to a predefined scheme.

Abstract translation: 音频处理装置包括多个电输入信号，每个电输入信号以数字形式提供，以及控制单元接收所述数字化电输入信号并提供所产生的增强信号。 控制单元被配置为根据预定义的方案从所述数字化电输入信号或从其导出的信号确定所得到的增强信号。

公开(公告)号：US09312830B1

公开(公告)日：2016-04-12

申请号：US14094323

申请日：2013-12-02

Applicant: Jeffrey Clark

Inventor： Jeffrey Clark

IPC: H03G7/00

CPC classification number: H03G7/002 ,  H03G3/3005

Abstract: An attenuation control before digital signal processing provides digital headroom. A pre-digital signal processing volume control receives a volume control signal and provides a pre-digital signal processing first attenuation of an input signal before digital signal processing. A master volume control provides a master volume control attenuation after the digital signal processing. The first attenuation may be the same as the master volume control attenuation, or it may be less or more than the master volume control attenuation, depending on the application and desired headroom. The first attenuation and master volume control attenuation may be linearly related or non-linearly related, or not a mathematical function (non-continuous, etc) at all. Changes in the attenuation may be ramped to avoid discontinuities in the signal, preferably over a period between 10 and 100 milliseconds, and are preferably applied as close to simultaneously with master volume changes as possible.

Abstract translation: 数字信号处理前的衰减控制提供了数字净空。 预数字信号处理音量控制接收音量控制信号，并在数字信号处理之前提供预数字信号处理输入信号的第一衰减。 主音量控制在数字信号处理之后提供主音量控制衰减。 第一衰减可以与主音量控制衰减相同，或者可以小于或大于主音量控制衰减，这取决于应用和期望的净空。 第一衰减和主音量控制衰减可以是线性相关的或非线性相关的，或者不是数学函数（非连续等））。 衰减的改变可以被斜坡化以避免信号中的不连续性，优选在10到100毫秒之间的时间段内，并且优选地应用尽可能接近于主音量变化的同时。

公开(公告)号：US09312826B2

公开(公告)日：2016-04-12

申请号：US14207252

申请日：2014-03-12

Applicant: KOPIN CORPORATION

Inventor： Dashen Fan

IPC: G10L21/0316 ,  H03G3/00 ,  H04R3/00 ,  G10L25/84 ,  H03G7/00 ,  G10L21/02 ,  G10L21/0216

CPC classification number: G10L21/0232 ,  G10K11/002 ,  G10K2210/3015 ,  G10K2210/3028 ,  G10L21/02 ,  G10L25/84 ,  G10L2021/02166 ,  G10L2025/783 ,  H03G3/00 ,  H03G7/002 ,  H03G7/007 ,  H04R3/005 ,  H04R2410/05 ,  H04R2460/01

Abstract: Systems and methods are described to automatically balance acoustic channel sensitivity. A long-term power level of a main acoustic signal is calculated to obtain an averaged main acoustic signal. Segments of the main acoustic signal are excluded from the averaged main acoustic signal using a desired voice activity detection signal. A long-term power level of a reference acoustic signal is calculated to obtain an averaged reference acoustic signal. Segments of the reference acoustic signal are excluded from the averaged reference acoustic signal using a desired voice activity detection signal. An amplitude correction signal is created using the averaged main acoustic signal and the averaged reference acoustic signal.

Abstract translation: 描述系统和方法以自动平衡声道灵敏度。 计算主声信号的长期功率电平以获得平均的主声信号。 使用期望的语音活动检测信号，从平均主声信号中排除主声信号的分段。 计算参考声学信号的长期功率电平以获得平均的参考声学信号。 使用期望的语音活动检测信号，从平均的参考声学信号中排除参考声学信号的分段。 使用平均的主声信号和平均的参考声信号来产生幅度校正信号。