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CN118633302A - Automatic Audio Tuning Launcher and Reporting

具体实施方式DETAILED DESCRIPTION

将容易理解的是，可以以各种不同的配置布置和设计在本文附图中总体描述和说明的即时组件。因此，对附图中所示的方法、装置、非暂态计算机可读介质和系统中的至少一者的实施例的以下详细描述不旨在限制所要求保护的申请的范围，而仅仅代表所选择的实施例。It will be readily appreciated that the instant components generally described and illustrated in the figures herein may be arranged and designed in a variety of different configurations. Therefore, the following detailed description of the embodiments of at least one of the methods, devices, non-transitory computer-readable media, and systems illustrated in the figures is not intended to limit the scope of the claimed application, but merely represents selected embodiments.

可以以任何合适的方式在一个或多个实施例中组合本说明书中描述的即时特征、结构或特性。例如，本说明书中的短语“示例实施例”、“一些实施例”或其他类似语言的使用是指结合该实施例描述的特定特征、结构或特性可以包括在至少一个实施例中的事实。因此，本说明书中的短语“示例实施例”、“在一些实施例中”、“在其他实施例中”或其他类似语言的出现不一定都指同一组实施例，并且可以以任何合适的方式在一个或多个实施例中组合所描述的特征、结构或特性。The instant features, structures, or characteristics described in this specification may be combined in any suitable manner in one or more embodiments. For example, the use of the phrases "example embodiments," "some embodiments," or other similar language in this specification refers to the fact that a particular feature, structure, or characteristic described in connection with the embodiment may be included in at least one embodiment. Thus, the appearance of the phrases "example embodiments," "in some embodiments," "in other embodiments," or other similar language in this specification does not necessarily refer to the same set of embodiments, and the described features, structures, or characteristics may be combined in any suitable manner in one or more embodiments.

另外，虽然在实施例的描述中使用了术语“消息”，但是本申请可以应用于许多类型的网络数据，例如分组、帧、数据报等。术语“消息”还包括分组、帧、数据报及其任何等同物。此外，虽然在示例实施例中描述了某些类型的消息和信令，但是它们不限于某一类型的消息，并且本申请不限于某一类型的信令。In addition, although the term "message" is used in the description of the embodiment, the present application can be applied to many types of network data, such as packets, frames, datagrams, etc. The term "message" also includes packets, frames, datagrams and any equivalents thereof. In addition, although certain types of messages and signaling are described in the example embodiments, they are not limited to a certain type of message, and the present application is not limited to a certain type of signaling.

用于为音频系统建立自动调谐和配置设置的启动过程可以包括一系列操作。在自动配置阶段，系统固件可以使用基于以太网的联网协议发现附接到中央控制器设备的外围设备。这些外围设备可以包括波束跟踪麦克风、放大器、通用串行总线(USB)和蓝牙(BT)I/O接口以及电话拨号盘设备。然后，设备固件修改其自身的配置和所发现的外围设备的配置，以将它们彼此相关联并且通过适当的音频信号处理功能路由相关联的音频信号。自动调谐阶段具有三个子阶段，麦克风(mic)和扬声器检测、调谐和验证。The startup process for establishing automatic tuning and configuration settings for the audio system can include a series of operations. In the automatic configuration phase, the system firmware can use an Ethernet-based networking protocol to discover peripheral devices attached to the central controller device. These peripheral devices can include beam tracking microphones, amplifiers, universal serial bus (USB) and Bluetooth (BT) I/O interfaces, and telephone dial devices. The device firmware then modifies its own configuration and the configuration of the discovered peripheral devices to associate them with each other and route the associated audio signals through appropriate audio signal processing functions. The automatic tuning phase has three sub-phases, microphone (mic) and speaker detection, tuning, and verification.

并不是由控制器设备管理的每个放大器输出声道(未示出)都可以具有附接的扬声器。在麦克风和扬声器检测阶段，从每个放大器声道顺序播放唯一的检测信号。在每个检测信号回放期间同时监视由所有麦克风检测到的输入信号。使用该技术，识别未连接的放大器输出声道，并且验证每个麦克风输入信号的完整性。在调谐阶段期间，从每个连接的放大器输出声道按顺序播放其他唯一的测试信号。这些信号再次由所有麦克风同时监视。具有对(一个或多个)麦克风的频率响应的先验知识，并且使用各种音频处理技术，固件可以计算房间的背景噪声水平和噪声频谱、每个放大器声道和连接的扬声器的灵敏度(针对给定的信号水平生成的房间SPL)、每个扬声器的频率响应、从每个麦克风到每个扬声器的距离、房间混响时间(RT60)等。使用这些计算，固件能够计算调谐参数以优化每个扬声器声道的水平设置，从而实现给定的目标SPL、每个扬声器声道的EQ设置，以既归一化扬声器的频率响应又实现目标房间频率响应。声学回声消除(AEC)、降噪(NR)和非线性处理(NLP)设置对于房间环境是最适当和有效的。Not every amplifier output channel (not shown) managed by the controller device can have an attached speaker. During the microphone and speaker detection phase, a unique detection signal is played sequentially from each amplifier channel. The input signal detected by all microphones is monitored simultaneously during each detection signal playback. Using this technology, unconnected amplifier output channels are identified and the integrity of each microphone input signal is verified. During the tuning phase, other unique test signals are played sequentially from each connected amplifier output channel. These signals are again monitored simultaneously by all microphones. With prior knowledge of the frequency response of (one or more) microphones, and using various audio processing techniques, the firmware can calculate the background noise level and noise spectrum of the room, the sensitivity of each amplifier channel and the connected speaker (the room SPL generated for a given signal level), the frequency response of each speaker, the distance from each microphone to each speaker, the room reverberation time (RT60), etc. Using these calculations, the firmware can calculate the tuning parameters to optimize the level setting of each speaker channel, thereby achieving a given target SPL, the EQ setting of each speaker channel, to normalize the frequency response of the speaker and achieve the target room frequency response. Acoustic echo cancellation (AEC), noise reduction (NR), and non-linear processing (NLP) settings are most appropriate and effective for the room environment.

在应用调谐参数之后发生验证阶段。在该阶段期间，测试信号从每个连接的放大器输出声道再次被按顺序播放并且由所有麦克风同时监视。测量用于验证系统达到目标SPL并且系统达到目标房间频率响应。在验证阶段期间，专门设计的语音清晰度测试信号被从所有扬声器播放并且由所有麦克风同时监视。语音清晰度是声音由收听者能够正确识别和理解的程度的行业标准量度。在信息报告中提供所进行的大多数测量和由自动设置应用的设置，以便从设备下载。The Verification Phase occurs after the tuning parameters are applied. During this phase, test signals are again played sequentially from each connected amplifier output channel and monitored by all microphones simultaneously. Measurements are used to verify that the system achieves the target SPL and that the system achieves the target room frequency response. During the Verification Phase, a specially designed speech intelligibility test signal is played from all speakers and monitored by all microphones simultaneously. Speech intelligibility is the industry standard measure of how well a sound can be correctly identified and understood by a listener. Most of the measurements made and the settings applied by Auto Setup are provided in an Information Report for download from the device.

示例实施例提供了一种系统，该系统包括用于管理多个麦克风和扬声器以在特定环境(例如，工作场所环境、会议室、会议厅、多个房间、不同楼层上的多个房间等)中提供音频优化调谐管理的控制器或中央计算机系统。音频系统的自动调谐包括调谐各种声级，执行均衡，识别目标声压级(SPL)，确定是否需要压缩，测量语音清晰度，确定最佳增益近似值以应用于扬声器/麦克风等。环境可以包括多个麦克风和扬声器区域，其中，各种扬声器通过不同的距离分隔开。第三方测试设备不理想并且不提供简化的可缩放性。理想的情况下，识别在网络上活动的网络组件并且仅使用这些组件设置用于会议或其他呈现目的的优化音频平台对于时间、专业知识和费用目的将是最佳的。Example embodiments provide a system that includes a controller or central computer system for managing multiple microphones and speakers to provide audio optimized tuning management in a specific environment (e.g., a workplace environment, a conference room, a meeting room, multiple rooms, multiple rooms on different floors, etc.). Automatic tuning of the audio system includes tuning various sound levels, performing equalization, identifying target sound pressure levels (SPLs), determining whether compression is needed, measuring speech intelligibility, determining optimal gain approximations to apply to speakers/microphones, etc. The environment may include multiple microphone and speaker zones, where various speakers are separated by different distances. Third-party test equipment is not ideal and does not provide simplified scalability. Ideally, identifying network components active on the network and using only those components to set up an optimized audio platform for conferencing or other presentation purposes would be optimal for time, expertise, and expense purposes.

自动均衡过程可以能够自动地将任何房间中的任何扩音器的频率响应均衡到可以由平直线和/或参数式曲线定义的任何期望的响应形状。该过程可能不在活动程序音频事件期间而在系统设置程序期间实时地操作。该过程考虑并且均衡对数幅度频率响应(分贝对频率)，并且可能不尝试均衡相位。该过程识别具有与所测量的响应的倒数非常匹配的频率响应的最佳滤波器，以便使曲线变平或重塑为一些其他的期望的响应值。该过程可以使用钟形单-双二阶无限冲激响应(IIR)滤波器来升压或切断参数式滤波器、低通和/或高通滤波器。还可以使用FIR滤波器，但是IIR滤波器具有优化的计算效率和低频分辨率，并且更适合于在房间中的宽广的收听区域进行空间平均或均衡。The automatic equalization process may be able to automatically equalize the frequency response of any loudspeaker in any room to any desired response shape that may be defined by a flat line and/or a parametric curve. The process may operate in real time during a system setup procedure, not during an active program audio event. The process considers and equalizes the logarithmic magnitude frequency response (decibels versus frequency), and may not attempt to equalize the phase. The process identifies the best filter with a frequency response that closely matches the inverse of the measured response in order to flatten or reshape the curve to some other desired response value. The process may use a bell-shaped single-biquad infinite impulse response (IIR) filter to boost or cut off parametric filters, low-pass and/or high-pass filters. FIR filters may also be used, but IIR filters have optimized computational efficiency and low-frequency resolution, and are more suitable for spatial averaging or equalization over a wide listening area in a room.

当执行均衡过程时，识别期望的目标频率响应。通常，这将是具有低频滚降和高频滚降的平坦响应，以避免设计将尝试实现来自(一个或多个)限频扩音器的不可实现结果的滤波器组。目标中频带响应不必是平坦的，并且该过程允许以双二阶滤波器阵列形式的任何任意目标频率响应。该过程还允许用户在任何自动调谐过程之前对要应用的总DSP滤波器组设置最大dB升压或某些切断极限值。When performing the equalization process, the desired target frequency response is identified. Typically, this will be a flat response with low frequency roll-off and high frequency roll-off to avoid designing a filter bank that will try to achieve an unachievable result from a limited frequency loudspeaker(s). The target mid-band response does not have to be flat, and the process allows for any arbitrary target frequency response in the form of a biquad filter array. The process also allows the user to set a maximum dB boost or certain cutoff limits on the total DSP filter bank to be applied prior to any auto-tuning process.

图1示出了根据示例实施例的受控扬声器和麦克风环境。参考图1，该图示出了音频控制环境112，该音频控制环境112可以具有任何数量的扬声器114和麦克风116，以经由自动调谐程序来检测音频、播放音频、重放音频、调整音频输出水平等。配置100可以包括通过空间、墙壁和/或地板分隔开的各种不同的区域130至160。控制器128可以与所有音频元件通信，并且可以包括计算机、处理器、被设置为用于接收和产生音频的软件应用等。在该示例中，啁啾响应测量技术可以用于通过扩音器的测量来获取频率响应。FIG1 illustrates a controlled speaker and microphone environment according to an example embodiment. Referring to FIG1 , an audio control environment 112 is illustrated that may have any number of speakers 114 and microphones 116 to detect audio, play audio, replay audio, adjust audio output levels, etc. via an automatic tuning program. The configuration 100 may include various different areas 130 to 160 separated by spaces, walls, and/or floors. A controller 128 may communicate with all audio elements and may include a computer, a processor, a software application configured to receive and generate audio, etc. In this example, a chirp response measurement technique may be used to obtain a frequency response through measurement of a loudspeaker.

关于设置过程，与控制器128通信的用户设备的用户界面前端的启动选项(自动设置+自动调谐)可以提供一种方式，以测试(一个或多个)房间、(一个或多个)扬声器和(一个或多个)麦克风的声音简档(sound profile)。网络发现可以用于寻找插入并包括在系统设备列表中的设备，并且向它们提供基准配置以在操作期间启动。可以在设备发现过程期间以图形格式实现音频系统，然后操作员可以拖放数据以获得更具定制性的体验或复位到出厂默认级别。如果系统未充分调谐到某一级别(level)，则可以生成警报，并且也可以通过发送到所有已知设备的测试信号来发现任何错误连接。Regarding the setup process, a startup option (auto setup + auto tune) on the user interface front end of a user device in communication with the controller 128 can provide a way to test the sound profile of the room(s), speaker(s), and microphone(s). Network discovery can be used to find devices that are plugged in and included in the system device list and provide them with a baseline configuration to start up during operation. The audio system can be implemented in a graphical format during the device discovery process, and the operator can then drag and drop data to obtain a more customized experience or reset to factory default levels. If the system is not sufficiently tuned to a certain level, an alarm can be generated, and any faulty connections can also be discovered through a test signal sent to all known devices.

音频环境通常包括各种组件和设备，例如麦克风、放大器、扩音器、DSP设备等。在安装之后，设备需要被配置以充当集成系统。软件应用可以用于配置由每个设备执行的某些功能。控制器或中央计算设备可以存储配置文件，该配置文件可以在安装过程期间被更新以包括新发现的音频简档。An audio environment typically includes various components and devices, such as microphones, amplifiers, loudspeakers, DSP devices, etc. After installation, the devices need to be configured to function as an integrated system. Software applications can be used to configure certain functions performed by each device. A controller or central computing device can store configuration files that can be updated during the installation process to include newly discovered audio profiles.

执行自动调谐过程的一个途径可以包括允许自动调谐处理在还包含定制DSP处理的设备上操作。为了启用该组合特征，代码将发现定制配置内的适当信号的注入和监视点。利用所识别的注入和监视点，任何所选择的DSP处理布局将自动兼容。自动调谐过程中的一些操作将从每个扬声器以一次一个发出测试信号，这增加了当存在许多扬声器时的总测量时间。其他操作可以包括在同时或重叠的时间段内从所有扬声器发出测试信号，并且对接收和处理的聚集声音执行测试过程。One approach to performing the automatic tuning process may include allowing the automatic tuning process to operate on a device that also includes custom DSP processing. To enable this combined feature, the code will find the injection and monitoring points of the appropriate signals within the custom configuration. With the identified injection and monitoring points, any selected DSP processing layout will be automatically compatible. Some operations in the automatic tuning process will send test signals from each speaker one at a time, which increases the total measurement time when there are many speakers. Other operations may include sending test signals from all speakers at the same time or in overlapping time periods, and performing a test process on the aggregate sound received and processed.

为了减少总测量时间，可以同时从每个扬声器播放不同的信号。提供混合信号的一些不同方式可以包括：每个扬声器产生一个特定的正弦波，其中，对每个不同的扬声器使用唯一的频率，播放短的音乐作品，其中，每个扬声器在音乐作品的混合中播放唯一的乐器，或者可以仅将频率不同的音调分别与每个扬声器配对。在大量的扬声器的情况下，可以使用具有多种打击乐器的歌曲，每个扬声器播放一种鼓声。任何其他多声道声音混合可以用于驱动动态和/或定制声音测试的过程。存在其他声音事件检测算法，其能够检测许多其他声音的混合中的声音的存在，这些算法在本测试分析程序中可能有用。自动调谐(auto-tune)可以是语音提示和从每个扬声器播放的测试信号的组合。测试信号用于收集关于系统中的放大器、扬声器和麦克风以及这些设备在声学空间中的放置的信息。To reduce the total measurement time, different signals can be played from each speaker at the same time. Some different ways of providing a mixed signal can include: each speaker generates a specific sine wave, where a unique frequency is used for each different speaker, playing a short musical piece, where each speaker plays a unique instrument in the mix of the musical piece, or only tones with different frequencies can be paired with each speaker respectively. In the case of a large number of speakers, a song with multiple percussion instruments can be used, with each speaker playing a drum sound. Any other multi-channel sound mixture can be used to drive the process of dynamic and/or customized sound testing. There are other sound event detection algorithms that are capable of detecting the presence of sounds in a mixture of many other sounds, which may be useful in this test analysis program. Auto-tune can be a combination of voice prompts and test signals played from each speaker. The test signal is used to collect information about the amplifiers, speakers and microphones in the system and the placement of these devices in the acoustic space.

还有其他信号可以使用，这些信号用于收集针对测试而收集的同一房间和设备信息。可以基于不同的目标决定使用不同的信号，例如使用声音悦耳的信号，其可以包括语音和/或音乐提示。优点是消除在空间中播放的科学声音测试音调。潜在的缺点是从不太理想的源信号中提取房间和设备信息所需的附加时间。为了减少总的测量时间，可以消除语音提示，并且可以使用产生最快结果的基本测试信号。There are other signals that can be used that are used to collect the same room and device information that is collected for testing. The decision to use different signals can be based on different goals, such as using a pleasing-sounding signal that can include speech and/or music cues. The advantage is the elimination of scientifically sound test tones played in the space. The potential disadvantage is the additional time required to extract the room and device information from the less-than-ideal source signals. To reduce the overall measurement time, the speech cues can be eliminated and the basic test signals that produce the fastest results can be used.

自动均衡程序(参见图3)能够自动地将任何房间中的任何扩音器的频率响应进行均衡，以达到可以由平直线和/或参数式曲线定义的任何期望的响应形状。该程序可能在系统设置程序期间是实时的，而不是在活动程序音频事件期间是实时的。该程序均衡对数幅度频率响应(分贝对频率)，并且可能不均衡相位。该程序识别一组最佳滤波器，该组最佳滤波器具有与所测量的响应的倒数非常匹配的频率响应，以使该响应变平或重塑为某个其他期望的响应值。该程序使用单-双二阶IIR滤波器，这些滤波器是钟型(例如，升压或切断参数式滤波器)、低通或高通滤波器。可以使用FIR滤波器，但是IIR滤波器具有更优的计算效率、低频分辨率，并且更适合于在房间中的宽广的收听区域进行空间平均和/或均衡。The automatic equalization program (see FIG. 3 ) can automatically equalize the frequency response of any loudspeaker in any room to any desired response shape that can be defined by a flat line and/or a parametric curve. The program may be real-time during the system setup program, rather than during an active program audio event. The program equalizes the logarithmic magnitude frequency response (decibels versus frequency) and may not equalize the phase. The program identifies a set of optimal filters that have a frequency response that closely matches the inverse of the measured response to flatten or reshape the response to some other desired response value. The program uses single-biquad IIR filters, which are bell-type (e.g., boosted or cut-off parametric filters), low-pass or high-pass filters. FIR filters can be used, but IIR filters have better computational efficiency, low-frequency resolution, and are more suitable for spatial averaging and/or equalization over a wide listening area in a room.

当执行均衡过程时，首先识别期望的目标频率响应。通常，这将是具有低频滚降(roll-off)和高频滚降的平坦响应，以避免该过程设计如下滤波器组：所述滤波器组将尝试实现来自限频扩音器的无法实现的结果。目标中频带响应(target mid-band response)不必是平坦的，并且该程序允许以双二阶滤波器阵列为形式的任何任意的目标频率响应。该程序还允许用户对要应用的总DSP滤波器组设置最大dB升压或削减限制。When performing the equalization process, the desired target frequency response is first identified. Typically, this will be a flat response with low frequency roll-off and high frequency roll-off to avoid the process designing a filter bank that will try to achieve unachievable results from a limited frequency loudspeaker. The target mid-band response does not have to be flat, and the program allows any arbitrary target frequency response in the form of a biquad filter array. The program also allows the user to set a maximum dB boost or cut limit on the total DSP filter bank to be applied.

与自动设置程序相关联的一个示例程序(参见图2)可以通过每个扬声器输出声道提供排序并且针对每个输出执行以下操作：逐步增强多音信号直到检测到期望的SPL水平，确定扬声器输出声道是否正常工作，确定所有麦克风(mic)输入声道是否正常工作，为测试信号设置未知放大器和扬声器的初步输出增益，测量来自所有麦克风的环境噪声以为RT60测量设置基础(RT60测量是对声音在具有扩散声场的空间中衰减60dB所花费的时间的测量)，以及检查过多噪声，提供啁啾测试信号，将来自所有‘N’个麦克风的啁啾响应同时记录到阵列中，对来自给出‘N’个脉冲响应的‘N’个麦克风的所有啁啾进行去卷积，并且针对每个麦克风输入：定位主脉冲峰值并且计算从扬声器到麦克风的距离，计算平滑的对数幅度频率响应并且应用麦克风补偿值(使用已知的麦克风灵敏度)，计算所有频率上的SPL平均值，对所有麦克风的频率响应求平均以获得空间平均值，对空间平均响应执行自动均衡以匹配目标响应，使用SPL水平以及最近的麦克风和最远的麦克风的距离计算房间衰减，使用来自最近的麦克风的SPL水平和房间衰减来计算输出增益以在距所有麦克风的平均距离处实现所需水平，计算SPL限制器阈值，其中启用自动EQ和自动增益，产生啁啾以测量和验证响应，测量每个麦克风的倍频带RT60，以及测量来自每个麦克风的平均SPL，然后对所有麦克风求平均以获得所实现的SPL水平。An example procedure associated with the automatic setup procedure (see Figure 2) may provide sequencing through each speaker output channel and for each output: gradually boost a multi-tone signal until a desired SPL level is detected, determine if the speaker output channels are functioning properly, determine if all microphone (mic) input channels are functioning properly, set preliminary output gain of the unknown amplifier and speaker for the test signal, measure ambient noise from all microphones to set the basis for RT60 measurements (the RT60 measurement is a measurement of the time it takes for a sound to decay by 60 dB in a space with a diffuse sound field), as well as check for excessive noise, provide a chirp test signal, record chirp responses from all ‘N’ microphones simultaneously into the array, deconvolute all chirps from the ‘N’ microphones giving ‘N’ impulse responses, and for each microphone input: determine The system detects the peak of the main impulse and calculates the distance from the speaker to the microphone, calculates a smoothed log magnitude frequency response and applies microphone compensation values (using known microphone sensitivities), calculates the SPL average over all frequencies, averages the frequency responses of all microphones to get a spatial average, performs automatic equalization on the spatial average response to match a target response, calculates the room attenuation using the SPL level and the distance of the nearest and farthest microphones, calculates the output gain using the SPL level from the nearest microphone and the room attenuation to achieve the desired level at the average distance from all microphones, calculates the SPL limiter threshold with auto EQ and auto gain enabled, generates chirp to measure and verify the response, measures the octave band RT60 for each microphone, and measures the average SPL from each microphone and then averages across all microphones to get the achieved SPL level.

另一示例实施例可以包括自动设置程序，该自动设置程序包括：确定哪些输入麦克风正在工作以及哪些输出扬声器声道正在工作，对每个输出扬声器声道执行自动均衡以达到任何期望的目标频率响应(由参数式EQ参数定义)，自动设置每个输出路径增益以实现由扬声器到麦克风的平均距离确定的房间中心处的目标SPL水平，自动设置输出限制器以达到房间中心处的最大SPL水平，基于房间测量自动设置自动回声消除(AEC)、非线性处理(NLP)和降噪(NR)值，测量房间中每个输出扬声器声道的频率响应，从每个输出声道测量在房间中心处的预期的最终标称SPL水平，测量房间的倍频带和全频带混响时间，测量每个麦克风的噪声频谱和倍频带噪声，测量房间的噪声标准(NC)评级，以及测量所有麦克风与扬声器的最小、最大和平均距离和房间的语音清晰度。所有测量数据可以用于建立最佳扬声器和麦克风配置值。Another example embodiment may include an automatic setup procedure that includes: determining which input microphones are operating and which output speaker channels are operating, performing automatic equalization on each output speaker channel to achieve any desired target frequency response (defined by parametric EQ parameters), automatically setting each output path gain to achieve a target SPL level at the center of the room determined by the average speaker-to-microphone distance, automatically setting output limiters to achieve a maximum SPL level at the center of the room, automatically setting automatic echo cancellation (AEC), non-linear processing (NLP), and noise reduction (NR) values based on room measurements, measuring the frequency response of each output speaker channel in the room, measuring the expected final nominal SPL level at the center of the room from each output channel, measuring the octave band and full-band reverberation time of the room, measuring the noise spectrum and octave band noise of each microphone, measuring the Noise Criteria (NC) rating of the room, and measuring the minimum, maximum, and average distances of all microphones from the speakers and the speech intelligibility of the room. All measurement data can be used to establish optimal speaker and microphone configuration values.

在一个示例音频系统设置程序中，用户界面上的启动操作(即，自动设置+自动调谐)可以提供启动对房间、扬声器和麦克风的声音简档的测试的方式。网络发现可以用于寻找插入并将包括在系统设备列表中的设备，并且向它们提供基准配置以在音频使用场景期间启动。可以在设备发现过程期间以图形格式来实现音频系统，操作员可以在自动系统配置之前或之后与显示器和拖放数据对接以获得更可定制的体验或复位到出厂默认级别。如果系统未充分调谐到某一级别，则可以生成警报并且也可以通过发送到所有已知设备的测试信号来发现任何错误连接。In an example audio system setup procedure, a startup operation on the user interface (i.e., Auto Setup + Auto Tune) can provide a way to start testing the sound profile of the room, speakers, and microphones. Network discovery can be used to find devices that are plugged in and will be included in the system device list and provide them with a baseline configuration to start during an audio usage scenario. The audio system can be implemented in a graphical format during the device discovery process, and the operator can interface with the display and drag and drop data before or after the automatic system configuration to obtain a more customizable experience or reset to factory default levels. If the system is not sufficiently tuned to a certain level, an alarm can be generated and any incorrect connections can also be discovered through a test signal sent to all known devices.

音频环境通常包括各种组件和设备，例如麦克风、放大器、扩音器、数字信号处理(DSP)设备等。在安装之后，设备需要被配置以充当集成系统。应用的软件可以用于配置由每个设备执行的某些功能。控制器或中央计算设备可以存储配置文件，该配置文件可以在安装过程期间被更新，以包括基于被安装的当前硬件的新发现的音频简档、(一个或多个)音频环境简档和/或期望的配置。在一个示例实施例中，自动调谐程序可以调谐包括由中央网络控制器管理的所有可访问硬件的音频系统。音频输入/输出水平、均衡和声压级(SPL)/压缩值都可以被选择用于特定环境中的最佳表现。The audio environment usually includes various components and equipment, such as microphones, amplifiers, loudspeakers, digital signal processing (DSP) equipment, etc. After installation, the equipment needs to be configured to act as an integrated system. The software of the application can be used to configure certain functions performed by each device. A controller or a central computing device can store a configuration file, which can be updated during the installation process to include newly discovered audio profiles, (one or more) audio environment profiles and/or desired configurations based on the current hardware installed. In an example embodiment, an automatic tuning program can tune an audio system including all accessible hardware managed by a central network controller. Audio input/output levels, equalization, and sound pressure level (SPL)/compression values can all be selected for optimal performance in a specific environment.

在自动设置期间，执行对哪些输入麦克风正在工作以及哪些输出扬声器声道正在工作的确定。对每个输出扬声器声道进行自动均衡，以达到期望的目标频率响应(由参数式EQ参数、高通滤波器、低通滤波器等定义)。默认选项可以是“平坦”响应。附加操作可以包括：自动设置每个输出路径增益，以实现在房间中心处的用户的目标SPL水平(假定麦克风的平均距离)，以及针对房间中心处的用户的最大SPL水平自动设置输出限制器。另一特征可以包括：基于房间测量自动确定自动回声消除(AEC)、非线性处理(NLP)和NRD值。还可以被执行的以下信息测量，包括：测量房间中每个输出扬声器声道的频率响应，从每个输出声道测量在房间中心处的预期的最终标称SPL水平，测量房间的倍频带混响时间(RT－60)，以及测量房间中的噪声底。附加特征可以包括测量所有麦克风与扬声器的最小、最大和平均距离。这些值可以提供执行附加自动设置(例如基于房间的较低频带中的混响时间来设置波束跟踪麦克风的高通滤波器截止频率)所需的信息，以及调谐AEC的自适应滤波器简档以最佳地匹配房间的期望回声特性。所获得的信息可以被保存在存储器中并且被应用使用，以提供会议室的声学特征和声音质量特性的示例。可以基于房间音频特性使用某些推荐以增加麦克风和扩音器之间的间距，或者，由于RT-60(所预测的语音清晰度的混响“评分”)过高而经由扬声器和麦克风对房间进行声学调整。During automatic setup, a determination is performed of which input microphones are operating and which output speaker channels are operating. Each output speaker channel is automatically equalized to achieve a desired target frequency response (defined by parametric EQ parameters, high pass filters, low pass filters, etc.). The default option may be a "flat" response. Additional operations may include automatically setting each output path gain to achieve a target SPL level for a user at the center of the room (assuming an average distance for the microphones), and automatically setting output limiters for a maximum SPL level for a user at the center of the room. Another feature may include automatically determining automatic echo cancellation (AEC), non-linear processing (NLP), and NRD values based on room measurements. The following information measurements may also be performed, including measuring the frequency response of each output speaker channel in the room, measuring the expected final nominal SPL level at the center of the room from each output channel, measuring the octave band reverberation time (RT-60) of the room, and measuring the noise floor in the room. Additional features may include measuring the minimum, maximum, and average distances of all microphones from the speakers. These values may provide the information needed to perform additional automatic setup, such as setting the high pass filter cutoff frequency of the beam tracking microphone based on the reverberation time in the lower frequency bands of the room, and tuning the adaptive filter profile of the AEC to best match the expected echo characteristics of the room. The information obtained may be saved in memory and used by the application to provide an example of the acoustic characteristics and sound quality characteristics of the conference room. Certain recommendations may be used to increase the spacing between the microphone and loudspeaker based on the room audio characteristics, or to make acoustic adjustments to the room via the speakers and microphones because the RT-60 (the reverberation "score" for predicted speech intelligibility) is too high.

音频设置过程可以包括一组操作，例如暂停任何类型的会议音频布局能力以及向自动设置应用提供输入(麦克风)和输出(扩音器)控制。从而参与自动设置的每个输出扩音器将产生被设计为捕获房间的声学特性的一系列“啁啾”和/或音调。在房间中产生的声音的数量与参与自动设置过程的输入和输出的数量直接相关。例如，在具有三个麦克风和两个扩音器的系统中，自动设置将执行以下动作：(——第一扩音器——)，扩音器1产生由麦克风1捕获的一系列声音，扩音器1产生由麦克风2捕获的一系列声音，并且扩音器1产生由麦克风3捕获的一系列声音；(下一个扩音器)，扩音器2产生由麦克风1捕获的一系列声音，扩音器2产生由麦克风2捕获的一系列声音，扩音器2产生由麦克风3捕获的一系列声音，并且在该过程完成之后，恢复常规的会议布局音频处理。基于自动设置处理来调整每个扩音器的增益和均衡，基于自动设置处理来调谐房间的AEC表现，基于自动设置处理来调谐房间的麦克风LPF，并且记录房间的声学特性。可选地，向用户呈现描述自动设置过程的结果的一些汇总数据。可能的是，如果发现有缺陷的麦克风或扩音器，或者如果在处理过程中捕获了不期望的高音量的声音(例如，街道噪声)，则在处理时自动设置可能“失败”。然后，自动设置将停止，并且如果是这种情况，则将提醒终端用户。此外，可以使用友好的自动设置语音与用户讨论当自动设置在整个过程中工作时所进行的操作。The audio setup process may include a set of operations, such as pausing any type of conference audio layout capabilities and providing input (microphone) and output (loudspeaker) controls to the automatic setup application. Each output loudspeaker participating in the automatic setup will thereby produce a series of "chirps" and/or tones designed to capture the acoustic characteristics of the room. The number of sounds produced in the room is directly related to the number of inputs and outputs participating in the automatic setup process. For example, in a system with three microphones and two loudspeakers, the automatic setup will perform the following actions: (--first loudspeaker--), loudspeaker 1 produces a series of sounds captured by microphone 1, loudspeaker 1 produces a series of sounds captured by microphone 2, and loudspeaker 1 produces a series of sounds captured by microphone 3; (next loudspeaker), loudspeaker 2 produces a series of sounds captured by microphone 1, loudspeaker 2 produces a series of sounds captured by microphone 2, loudspeaker 2 produces a series of sounds captured by microphone 3, and after the process is completed, the conventional conference layout audio processing is resumed. The gain and equalization of each loudspeaker are adjusted based on the auto setup process, the AEC performance of the room is tuned based on the auto setup process, the microphone LPF of the room is tuned based on the auto setup process, and the acoustic characteristics of the room are recorded. Optionally, some summary data describing the results of the auto setup process is presented to the user. It is possible that the auto setup may "fail" during the process if a defective microphone or loudspeaker is found, or if an unexpectedly high volume of sound (e.g., street noise) is captured during the process. The auto setup will then stop and the end user will be alerted if this is the case. In addition, a friendly auto setup voice can be used to discuss with the user what the auto setup is doing as it works through the process.

图2示出了自动均衡过程，该自动均衡过程包括针对环境中的多个扬声器的迭代过程。参考图2，在启动程序期间，可以使用用户界面控制启动和“自动调谐”选项。可以执行存储器分配操作以检测某些扬声器、麦克风等。可以在存储器中存储所识别的网络元件。还可以执行使图2的操作启动的调谐程序。每个扬声器可以接收作为输入204的输出信号202以产生声音或信号。环境噪声水平也可以从扬声器被识别206并且由麦克风检测。可以向各个扬声器发送多个音调208，这些音调被测量并且值被存储在存储器中。此外，啁啾响应210可以用于确定扬声器的水平和对应的房间/环境。可以识别脉冲响应212，并且可以基于输入计算对应的频率响应值214。此外，可以计算语音清晰度评级(语音传输指数(STI))以及‘RT60’值，该‘RT60’值是对声音在具有扩散声场的空间中衰减60dB所花费的时间的测量，这意味着房间足够大，使得来自源的反射从所有方向以相同的水平到达麦克风。可以确定输入值的平均值216以估计对应网络元件的总声音值。求平均值可以包括对输入值的值求和并除以输入值的数量。FIG. 2 shows an automatic equalization process that includes an iterative process for multiple speakers in an environment. Referring to FIG. 2 , during the startup procedure, a user interface can be used to control the startup and “auto-tune” options. A memory allocation operation can be performed to detect certain speakers, microphones, etc. The identified network elements can be stored in the memory. A tuning program that enables the operation of FIG. 2 can also be performed. Each speaker can receive an output signal 202 as an input 204 to produce a sound or signal. The ambient noise level can also be identified 206 from the speaker and detected by the microphone. Multiple tones 208 can be sent to each speaker, which are measured and the values are stored in the memory. In addition, a chirp response 210 can be used to determine the level of the speaker and the corresponding room/environment. An impulse response 212 can be identified, and a corresponding frequency response value 214 can be calculated based on the input. In addition, a speech intelligibility rating (speech transmission index (STI)) and an 'RT60' value, which is a measure of the time it takes for sound to decay by 60 dB in a space with a diffuse sound field, may be calculated, meaning that the room is large enough so that reflections from the source arrive at the microphone at the same level from all directions. An average 216 of the input values may be determined to estimate the total sound value of the corresponding network element. Averaging may include summing the values of the input values and dividing by the number of input values.

继续相同的示例，可以基于输入响应的空间平均值来执行自动均衡218。可以输出自动均衡级别222，直到程序完成224。当完成输出224时，设置输出值226，该输出值可以包括在向各个扬声器输出音频信号时使用的参数。在验证程序230期间迭代地继续该过程，针对每个扬声器，该验证程序230可以包括类似的操作，例如202、204、210、212、214、216。此外，在迭代验证过程中，可以执行语音清晰度的测量，直到识别出所有输出值。如果在操作224中输出未完成，则使用自动均衡级别225继续处理下一个扬声器的下一输出值(即，迭代地)，并且继续直到测量并存储所有扬声器输出。Continuing with the same example, automatic equalization 218 can be performed based on the spatial average of the input response. The automatic equalization level 222 can be output until the program is completed 224. When the output 224 is completed, the output value 226 is set, which can include parameters used when outputting audio signals to each speaker. The process continues iteratively during the verification procedure 230, which can include similar operations, such as 202, 204, 210, 212, 214, 216 for each speaker. In addition, during the iterative verification process, a measurement of speech intelligibility can be performed until all output values are identified. If the output is not completed in operation 224, the next output value of the next speaker is processed using the automatic equalization level 225 (i.e., iteratively), and continue until all speaker outputs are measured and stored.

自动设置操作依赖于使用啁啾信号和可能的啁啾去卷积来测量扩音器、麦克风和房间参数以获得脉冲响应。可以使用啁啾信号去卷积来获取质量脉冲响应(IR)，该质量脉冲响应使用实际的FFT尺寸而没有噪声、系统失真和表面反射。将影响自动设置程序的有效性的一个项目是对诸如麦克风、功率放大器和扩音器之类的系统组件的已知程度。每当组件频率响应是已知时，数字信号处理器(DSP)应当在生成和记录任何啁啾信号之前应用校正均衡，以便增加啁啾测量的精确性。The auto setup operation relies on measuring loudspeaker, microphone and room parameters using a chirp signal and possible chirp deconvolution to obtain an impulse response. The chirp signal deconvolution can be used to obtain a quality impulse response (IR) that uses a practical FFT size without noise, system distortion and surface reflections. One item that will affect the effectiveness of the auto setup procedure is how well the system components such as microphones, power amplifiers and loudspeakers are known. Whenever the component frequency responses are known, the digital signal processor (DSP) should apply corrective equalization before generating and recording any chirp signals in order to increase the accuracy of the chirp measurement.

可以使用自动均衡程序来将任何房间中的任何扩音器的频率响应值均衡到期望的响应形状(例如，水平线和/或参数式曲线)。这种程序可以利用钟形类型的单-双二阶IIR滤波器。该过程可以从具有低频滚降和高频滚降的期望目标频率响应开始，以避免遇到为特定扩音器和房间建立的滤波器的限制。目标响应(H_target)可以是平坦的，具有低频滚降。通过使用啁啾刺激/响应测量，可以获得房间中扩音器的所测量的频率响应。响应需要被归一化以具有0dB的平均值，可以使用高频和低频极限值来均衡和设置所使用的数据的极限值。该程序将计算极限值之间的平均水平，并且从所测量的响应中减去该平均水平值以提供归一化为‘0’的响应(H_meas)。然后，通过从目标响应减去所测量的响应来确定限频目标滤波器：H_targfilt＝H_target-H_meas，并且该值是用于下一个自动EQ双二阶滤波器的目标响应。An automatic equalization program can be used to equalize the frequency response value of any loudspeaker in any room to a desired response shape (e.g., a horizontal line and/or a parametric curve). Such a program can utilize a single-biquad IIR filter of the bell type. The process can start with a desired target frequency response with a low frequency roll-off and a high frequency roll-off to avoid running into the limitations of the filters established for a particular loudspeaker and room. The target response (H _target ) can be flat with a low frequency roll-off. By using a chirp stimulus/response measurement, the measured frequency response of the loudspeaker in the room can be obtained. The response needs to be normalized to have a mean value of 0 dB, and high and low frequency limits can be used to equalize and set the limits of the data used. The program will calculate the average level between the limits and subtract this average level from the measured response to provide a response normalized to '0' (H _meas ). The frequency-limited target filter is then determined by subtracting the measured response from the target response: H _targfilt =H _target -H _meas , and this value is the target response for the next automatic EQ biquad filter.

为了找到参数式滤波器以拟合H_targfilt的曲线，通过被称为FindFreqFeatures()的函数来寻找所有重要的曲线特征(0dB交叉点和峰值点)。In order to find a parametric filter to fit the curve of H _targfilt , all important curve features (0 dB crossing points and peak points) are found by a function called FindFreqFeatures().

对两个频率极限值处的滤波器选择的处理稍微不同。如果目标滤波器在频率极限值处要求升压，则使用PEQ升压滤波器，其中心频率为极限值频率。如果目标滤波器在频率极限值处要求衰减(这通常在目标响应具有滚降时发生)，则选择HPF/LPF并且计算-3dB角频率以匹配到曲线为-3dB的点。当超出自动EQ范围时，尤其是需要滚降响应时，发现这能产生更好的匹配。一旦识别出目标滤波器的所有频率特征，就使用被称为FindBiggestArea()的函数来寻找目标的最显著的双二阶滤波器，该最显著的双二阶滤波器仅以如下所示的目标滤波器曲线下的最大面积为特性。The filter selection at the two frequency extremes is handled slightly differently. If the target filter requires a boost at the frequency extreme, a PEQ boost filter is used with its center frequency at the extreme frequency. If the target filter requires attenuation at the frequency extreme (this usually occurs when the target response has a roll-off), the HPF/LPF is selected and the -3dB corner frequency is calculated to match to the point where the curve is -3dB. This has been found to produce a better match when the auto EQ range is exceeded, especially when a roll-off response is required. Once all the frequency characteristics of the target filter are identified, a function called FindBiggestArea() is used to find the most significant biquad filter for the target, which is characterized by only the largest area under the target filter curve as shown below.

基于这些特性，被称为DeriveFiltParamsFromFreqFeatures()的函数基于曲线中心频率、dB升压/削减和带宽(Q)来计算3个参数(fctr、dB、Q)。2-极带通滤波器的带宽被定义为fctr/(f_upper-f_lower)，其中f_upper和f_lower是线性幅度是.707*峰值的位置。本文存在1+带通的钟形滤波器，但是根据经验发现，使用.707*峰值(dB)(其中基准是0dB)也提供了用于估计钟形的Q的最佳结果。边缘频率不用于计算PEQ带宽，而是用于描绘两个相邻的PEQ峰值。如果该区域表示在频率处的衰减，则该函数将计算LPF/HPF滤波器的角频率，其中响应是-3dB。根据这些滤波器参数，计算自动EQ双二阶滤波器系数，并且将双二阶添加到自动EQDSP滤波器组。然后，将该更新的DSP滤波器响应(H_dspfilt)添加到所测量的响应(H_meas){所有数量以dB为单位}，以示出自动均衡响应看起来是什么(H_autoeq)。然后，从目标响应(H_target)减去自动均衡响应(H_autoeq)以产生新的目标滤波器(H_targfilt)。该新的目标滤波器表示误差，即期望的目标响应和校正响应之间的差异。Based on these characteristics, a function called DeriveFiltParamsFromFreqFeatures() calculates 3 parameters (fctr, dB, Q) based on the curve center frequency, dB boost/cut, and bandwidth (Q). The bandwidth of a 2-pole bandpass filter is defined as fctr/(f _upper -f _lower ), where f _upper and f _lower are the locations where the linear amplitude is .707*peak. A bell filter with 1+bandpass is present in this article, but it has been empirically found that using .707*peak (dB) (where the reference is 0dB) also provides the best results for estimating the Q of the bell. The edge frequencies are not used to calculate the PEQ bandwidth, but are used to delineate two adjacent PEQ peaks. If this area represents attenuation at a frequency, the function will calculate the corner frequency of the LPF/HPF filter where the response is -3dB. Based on these filter parameters, the auto EQ biquad filter coefficients are calculated, and the biquad is added to the auto EQ DSP filter bank. This updated DSP filter response (H _dspfilt ) is then added to the measured response (H _meas ) {all quantities in dB} to show what the auto-equalization response looks like (H _autoeq ). The auto-equalization response (H _autoeq ) is then subtracted from the target response (H _target ) to produce a new target filter (H _targfilt ). This new target filter represents the error, i.e., the difference between the desired target response and the correction response.

图3示出了根据示例实施例的用于确定自动均衡滤波器组以应用于扩音器环境的过程。参考图3，该过程可以包括将目标响应定义为双二阶滤波器和HPF/LPF频率列表302，测量来自麦克风的啁啾响应304，将该值归一化为频率极限值之间的0dB 306，从目标响应减去所测量的响应以提供目标滤波器308，寻找目标滤波器过零点和导数零点310，将两组零频率按顺序组合以识别频率特征值312，识别目标滤波器曲线下方的最大区域314，导出参数以拟合在.707处的频率乘以峰值的钟形面积316，确定滤波器参数是否是可听见的318，如果是，则过程继续以基于所识别的滤波器参数计算双二阶系数320。该过程继续以基于幅度极限值限制滤波器dB 322，将该新的被限制的滤波器添加到DSP滤波器组324，将未被限制的EQ滤波器添加到所测量的响应以提供未被限制的校正响应326，以及从目标响应减去该校正响应以提供新的目标滤波器328。如果使用了所有可用的二阶330，则过程结束322，或者如果否，则过程继续回到操作310。FIG3 illustrates a process for determining an automatic equalization filter bank for application to a loudspeaker environment according to an example embodiment. Referring to FIG3, the process may include defining a target response as a biquad filter and a list of HPF/LPF frequencies 302, measuring a chirp response from a microphone 304, normalizing the value to 0 dB between frequency extremes 306, subtracting the measured response from the target response to provide a target filter 308, finding target filter zero crossings and derivative zeros 310, combining the two sets of zero frequencies in order to identify frequency eigenvalues 312, identifying the maximum area under the target filter curve 314, deriving parameters to fit the frequency at .707 times the bell-shaped area of the peak 316, determining if the filter parameters are audible 318, and if so, the process continues with calculating biquad coefficients 320 based on the identified filter parameters. The process continues by limiting the filter dB based on the amplitude limit value 322, adding the new limited filter to the DSP filter bank 324, adding the unlimited EQ filter to the measured response to provide an unlimited correction response 326, and subtracting the correction response from the target response to provide a new target filter 328. If all available second orders are used 330, the process ends 322, or if not, the process continues back to operation 310.

为了确定哪个扩音器(扬声器)输出是实时的，五倍频带多音(五个正弦波信号间隔一个倍频带)信号水平被施加到扬声器，并且以快速的速率逐步增加，以便快速检测任何连接的实时的扬声器。多音信号水平每次增加一个扬声器，同时监视所有麦克风的信号水平。一旦一个麦克风(mic)以期望的音频系统声压级(SPL)目标水平(即，SPL阈值水平)接收到信号，则多音测试信号被终止，并且扬声器输出声道被指定为是实时的。如果多音测试信号达到最大‘安全极限值’并且没有麦克风接收到目标SPL水平，则扬声器输出被指定为无效的/断开的。所接收的五倍频带信号通过一组五个窄带通滤波器。五个倍频带测试音调和五个带通滤波器的目的是防止来自宽带环境噪声或从房间中的一些其他源产生的单个音调的对扬声器的错误检测。换句话说，音频系统正在产生和接收特定的信号签名(signature)，以将该信号与房间中的其他无关声源区分开。用于检测有效扬声器输出的同一个五倍频带多音同时用于检测有效麦克风输入。一旦最高麦克风信号达到音频系统目标SPL水平，则多音测试信号被终止。此时，所有的麦克风信号水平被记录。如果麦克风信号高于某个最小阈值水平，则麦克风输入被指定为实时的麦克风输入，否则其被指定为无效的/断开的。In order to determine which loudspeaker (speaker) output is real-time, a five-octave band multi-tone (five sine wave signals are spaced one octave band) signal level is applied to the speaker and gradually increased at a fast rate to quickly detect any connected real-time speakers. The multi-tone signal level increases one speaker at a time while monitoring the signal levels of all microphones. Once a microphone (mic) receives a signal at the desired audio system sound pressure level (SPL) target level (i.e., SPL threshold level), the multi-tone test signal is terminated and the speaker output channel is designated as real-time. If the multi-tone test signal reaches the maximum 'safe limit value' and no microphone receives the target SPL level, the speaker output is designated as invalid/disconnected. The received five-octave band signal passes through a set of five narrow bandpass filters. The purpose of the five octave band test tones and five bandpass filters is to prevent false detection of speakers from single tones generated from broadband ambient noise or some other sources in the room. In other words, the audio system is generating and receiving a specific signal signature to distinguish the signal from other irrelevant sound sources in the room. The same five-octave band multi-tone used to detect valid speaker output is also used to detect valid microphone input. Once the highest microphone signal reaches the audio system target SPL level, the multi-tone test signal is terminated. At this point, all microphone signal levels are recorded. If the microphone signal is above a certain minimum threshold level, the microphone input is designated as a live microphone input, otherwise it is designated as invalid/disconnected.

为了设置扩音器输出增益水平，以dB为单位的所期望的SPL声学收听水平将被确定并被存储在固件中。DSP扩音器输出声道将其增益设置为实现该目标SPL水平。如果功率放大器增益是已知的，并且扩音器灵敏度是已知的，则可以针对特定SPL水平来准确地设置这些输出DSP增益，例如基于距每个扩音器一米(考虑其他距离并且可以将其作为替代)。在某些估计的收听者位置处的水平将是小于该估计水平的某个水平。在自由空间中，与源的距离每增加一倍，声级下降6dB。对于典型的会议室，与源的距离增加一倍时的水平可以被识别为-3dB。如果假设每个收听者将在离最近的扩音器2米到8米的范围内，并且针对4米的中间距离设置增益，则所得的声学等级将在期望水平的+/-3dB内。如果(一个或多个)扩音器的灵敏度是未知的，则将使用从最近的麦克风获得的啁啾响应信号。使用最近的麦克风的原因是为了最小化由于估计水平损失相对于距离的反射和误差。可以从该响应的水平和飞行时间(time-of-flight，TOF)估计扩音器灵敏度，尽管由于扩音器离轴拾取引起的衰减是未知的。如果功率放大器增益是未知的，则将使用29dB的典型值，这可能引入+/-3dB的SPL水平误差。In order to set the loudspeaker output gain level, the desired SPL acoustic listening level in dB will be determined and stored in the firmware. The DSP loudspeaker output channel sets its gain to achieve the target SPL level. If the power amplifier gain is known and the loudspeaker sensitivity is known, these output DSP gains can be accurately set for a specific SPL level, for example based on one meter from each loudspeaker (other distances are considered and can be used as an alternative). The level at some estimated listener position will be a level less than the estimated level. In free space, the sound level drops by 6dB for every doubling of the distance from the source. For a typical conference room, the level when the distance from the source is doubled can be identified as -3dB. If it is assumed that each listener will be within a range of 2 meters to 8 meters from the nearest loudspeaker, and the gain is set for an intermediate distance of 4 meters, the resulting acoustic level will be within +/-3dB of the desired level. If the sensitivity of (one or more) loudspeakers is unknown, the chirp response signal obtained from the nearest microphone will be used. The reason for using the nearest microphone is to minimize reflections and errors due to estimated level loss relative to distance. The microphone sensitivity can be estimated from the level and time-of-flight (TOF) of this response, although the attenuation due to off-axis pickup of the microphone is unknown. If the power amplifier gain is unknown, a typical value of 29dB will be used, which may introduce a +/-3dB SPL level error.

分析电声系统以识别应当用于实现最佳声学等级的增益。可以从任何声音系统导出电压、功率和声学等级以及增益。这些值可以用于使用DSP处理器提供在某一特定位置处的SPL水平。通常，音频系统将具有麦克风、扩音器、编解码器、DSP处理器和放大器。Analyze the electroacoustic system to identify the gain that should be used to achieve the best acoustic level. Voltage, power and acoustic level and gain can be derived from any sound system. These values can be used to provide the SPL level at a specific location using a DSP processor. Typically, an audio system will have a microphone, loudspeaker, codec, DSP processor and amplifier.

图4示出了根据示例实施例的用于识别各种音频信号水平和特性的示例配置。参考图4，该示例包括特定的房间或环境，例如一个会议室，该会议室里有人员436，人员436估计距离扩音器434约一米。衰减值被表示为增益值。例如，G_PS＝L_P-L_SPKR是来自扩音器的在距人员一米处的增益，例如，其可以是约-6dB。L_P是在不考虑任何特定的平均的情况下的声学声压级，L_SPKR是距扬声器1米处的声压值。G_MP是从麦克风432到人员的增益，G_MS是从麦克风到扩音器的增益。功率放大器424可以用于为麦克风供电，DSP处理器422可以用于接收和处理来自麦克风的数据以识别施加到扬声器434的最佳增益和功率水平。识别这些最佳值将理想地包括确定G_PS和G_PS。这将有助于实现在收听者位置处的声级以及设定的DSP输出增益和输入前置放大器增益值。FIG. 4 shows an example configuration for identifying various audio signal levels and characteristics according to an example embodiment. Referring to FIG. 4 , the example includes a specific room or environment, such as a conference room, in which there is a person 436, who is estimated to be about one meter away from a loudspeaker 434. The attenuation value is expressed as a gain value. For example, G _PS = L _P - L _SPKR is the gain from the loudspeaker at one meter away from the person, for example, it can be about -6 dB. L _P is the acoustic sound pressure level without considering any specific average, and L _SPKR is the sound pressure value at 1 meter away from the speaker. G _MP is the gain from the microphone 432 to the person, and G _MS is the gain from the microphone to the loudspeaker. The power amplifier 424 can be used to power the microphone, and the DSP processor 422 can be used to receive and process data from the microphone to identify the optimal gain and power level applied to the loudspeaker 434. Identifying these optimal values will ideally include determining G _PS and G _PS . This will help achieve the sound level at the listener position and the set DSP output gain and input preamplifier gain values.

在图4的这个示例中，如果知道关于麦克风、放大器和扩音器的几个基本参数，则L_{sens,mic,(1)PA}(dBu)是模拟麦克风的灵敏度，单位为dBu，作为相对于1帕斯卡(PA)的绝对量，在该示例中是-26.4dBu，G_amp是功率放大器的增益，在该示例中是29dB，而L_sens,spkr是扩音器的灵敏度，在该示例中是是90dBa。继续该示例，L_gen是信号生成器的水平(dBu)，G_dsp,in是包括麦克风前置放大器增益的DSP处理器输入的增益，在该示例中是54dB，G_dsp,out是DSP处理器输出增益的增益，在该示例中是-24dB。播放刺激信号，并且测量响应信号，其可以是例如14.4dBu，并且L_1PA＝94。在该示例中，可以通过L_mic＝L_dsp-L_sens,mic,1PA+L_1PA-G_dsp.in＝14.4-(-26.4)+94＝80.8dBa来识别麦克风处的声级。距扩音器1米处的声级是L_spkr＝L_gen+G_dsp+G_amp+L_sens,spkr-L_{sens,spkr,volts}＝0+(-24dB)+29dB+90dBa-11.3dBu＝83.7dBu。现在可以计算G_MS＝L_mis-L_spkr＝-2.9dBa。估计值以典型会议室内每增加一倍距离为-2.5dB为基础。In this example of FIG. 4 , if a few basic parameters about the microphone, amplifier, and loudspeaker are known, L _{sens,mic,(1)PA} (dBu) is the sensitivity of the analog microphone in dBu as an absolute quantity relative to 1 Pascal (PA), in this example -26.4 dBu, G _amp is the gain of the power amplifier, in this example 29 dB, and L _sens,spkr is the sensitivity of the loudspeaker, in this example 90 dBa. Continuing with this example, L _gen is the level of the signal generator (dBu), G _dsp,in is the gain of the DSP processor input including the microphone preamplifier gain, in this example 54 dB, and G _dsp,out is the gain of the DSP processor output gain, in this example -24 dB. The stimulus signal is played and the response signal is measured, which may be, for example, 14.4 dBu, and L _1PA =94. In this example, the sound level at the microphone can be identified by L _mic = L _dsp - L _sens,mic,1PA + L _1PA - G _dsp.in = 14.4 - (-26.4) + 94 = 80.8 dBa. The sound level at 1 meter from the loudspeaker is L _spkr = L _gen + G _dsp + G _amp + L _sens,spkr - L _{sens,spkr,volts} = 0 + (-24 dB) + 29 dB + 90 dBa - 11.3 dBu = 83.7 dBu. Now G _MS can be calculated = L _mis - L _spkr = -2.9 dBa. The estimate is based on -2.5 dB per doubling of distance in a typical conference room.

在麦克风、功率放大器和扩音器的增益和其他参数是未知的情况下，对于麦克风，L_p和L_mic的测量值通常是-38dBu，其中，对于功率放大器是+/-12dB、29dB+/-3dB，对于扩音器是90dBa+/-5dB。上述公式是计算期望声级的DSP增益和实现动态范围所必需的。然后，可以通过各种增益测量来识别期望的收听者水平L_P。In the case where the gain and other parameters of the microphone, power amplifier and loudspeaker are unknown, the measured values of _Lp and _Lmic are typically -38dBu for the microphone, +/-12dB, 29dB+/-3dB for the power amplifier, and 90dBa+/-5dB for the loudspeaker. The above formula is necessary to calculate the DSP gain and dynamic range for the desired sound level. The desired listener level _Lp can then be identified through various gain measurements.

图5示出了根据示例实施例的用于识别受控扬声器和麦克风环境中的声压级(SPL)的过程。参考图5，该示例包括模拟模型中的收听者436，该收听者436与特定房间中的扬声器534的距离是D_P。在自由空间中，距离每增加一倍，声学等级衰减是6dB。然而，在房间中，由于反射和混响，该衰减水平将是小于6dB的某个值。会议室中的声学等级衰减的典型值是距离每增加一倍，衰减约为3dB，其中，一般而言，小房间和/或反射性房间将比这一数值小一些，而大房间和/或吸收性房间将比这一数值大。FIG5 illustrates a process for identifying the sound pressure level (SPL) in a controlled speaker and microphone environment according to an example embodiment. Referring to FIG5 , the example includes a listener 436 in a simulation model, the listener 436 being at a distance D _P from a speaker 534 in a particular room. In free space, the acoustic level attenuation is 6 dB for each doubling of the distance. However, in a room, due to reflections and reverberation, the attenuation level will be some value less than 6 dB. A typical value for the acoustic level attenuation in a conference room is about 3 dB for each doubling of the distance, where, in general, small rooms and/or reflective rooms will be somewhat less than this value, while large rooms and/or absorptive rooms will be greater than this value.

在特定位置处使用多个麦克风在离扩音器534某个距离D_P处以某个期望收听者水平L_P产生期望的SPL、在距扩音器534一米处产生已知水平L₁，并且知道距离每增加一倍的衰减和扩音器的灵敏度。可以从如D1和D2所示的两个同时测量位置处的一个啁啾来确定所有这些参数。假设房间的水平均匀地衰减，则可以从房间中的任何两个测量值(在两个不同位置处)计算距离每增加一倍的衰减。当房间尺寸增加和/或变得更加扩散时，该假设更有效。该假设作为所有频率上的平均衰减也更有效。距离每增加一倍时的衰减等式可以被导出为：α_dd＝-(L₁-L₂)/log2(D₂/D₁)，其中，L＝SPL水平，D＝距离，而α_dd在该示例中是负值，在该示例中，衰减值被认为是负增益。距扩音器的位置L₁和L₂可以是任何顺序(即，不必是D2>D1)。接下来，必须测量扩音器灵敏度，该灵敏度是当由给定参考电压驱动时距扬声器‘1’米处的SPL水平。如果在距扬声器并非1m的某个距离处进行测量，则将通过使用α_dd和相对于1m的“距离加倍”(“doublings of distance”)来计算距扬声器1m处的水平。可以使用表达式OneMeterDoublings＝log2(D₁)来计算1m的距离加倍。现在可以使用L_1m＝L₁-OneMeterDoublings*αdd来计算将在1m处出现的水平。如果所使用的电测试信号是扬声器的灵敏度电参考水平，通常是2.83V(8欧姆时为1W)，则L_1m＝L_sens,spkr。然而，如果扬声器驱动电压有些不同，则可以简单地使用等式L_sens,spkr＝L_1m-L_dsp,FSout-G_dsp,out-G_amp-G_attn,out+L_{sens,spkr,volts}来计算L_sens,spkr。L_sens,spkr是扩音器的灵敏度，L_dsp,FSout是DSP处理器输出的灵敏度，G_dsp,out是DSP输出的增益，G_amp是功率放大器的增益，G_attn,out是任何衰减器的增益，L_{sens,spkr,volts}是扩音器的灵敏度(单位为伏特)。Multiple microphones are used at specific locations to produce a desired SPL at some desired listener level _LP at some distance _DP from the loudspeaker 534, a known level _L1 at one meter from the loudspeaker 534, and knowing the attenuation per doubling of distance and the sensitivity of the loudspeaker. All of these parameters can be determined from one chirp at two simultaneous measurement locations as shown by D1 and D2. Assuming that the room level is uniformly attenuated, the attenuation per doubling of distance can be calculated from any two measurements in the room (at two different locations). This assumption is more valid as the room size increases and/or becomes more diffuse. This assumption is also more valid as an average attenuation over all frequencies. The attenuation equation for each doubling of distance can be derived as: α _dd =-(L ₁ -L ₂ )/log2(D ₂ /D ₁ ), where L=SPL level, D=distance, and α _dd is a negative value in this example, in which the attenuation value is considered to be a negative gain. The positions _L1 and _L2 from the loudspeaker can be in any order (i.e. it is not necessary that D2>D1). Next, the loudspeaker sensitivity must be measured, which is the SPL level at '1' meter from the loudspeaker when driven by a given reference voltage. If the measurement is taken at some distance from the loudspeaker other than 1m, the level at 1m from the loudspeaker will be calculated by using _αdd and the "doublings of distance" relative to 1m. The doublings of distance for 1m can be calculated using the expression OneMeterDoublings=log2( _D1 ). The level that will appear at 1m can now be calculated using _L1m = _L1 -OneMeterDoublings*αdd. If the electrical test signal used is the loudspeaker's sensitivity electrical reference level, typically 2.83V (1W at 8 ohms), then _L1m =Lsens _,spkr . However, if the speaker drive voltage is somewhat different, then L _sens,spkr can be simply calculated using the equation L _sens,spkr = L _1m - L _dsp,FSout - G _dsp,out - G _amp - G _attn,out + L _{sens,spkr,volts} , where L _sens,spkr is the sensitivity of the loudspeaker, L _dsp,FSout is the sensitivity of the DSP processor output, G _dsp,out is the gain of the DSP output, G _amp is the gain of the power amplifier, G _attn,out is the gain of any attenuator, and L _{sens,spkr,volts} is the sensitivity of the loudspeaker in volts.

既然已经识别了房间和扬声器灵敏度的α_dd，在收听者距离D_P处产生期望水平L_P所需的扬声器驱动水平(或DSP输出增益)，可以通过计算到收听者位置的一米的加倍来确定：OneMeterDoublings＝log2(D₁)。接下来，可以计算距扩音器1m处的收听者水平：L_1m＝L₁-OneMeterDoublings*α_dd。最后，可以通过G_dsp，out＝L_1m-L_sens,spkr-L_dsp,FSout-G_amp-G_attnout+L_{sens,spkr,volts}来识别扩音器驱动水平或DSP输出增益。Now that the room and speaker sensitivity α _dd has been identified, the speaker drive level (or DSP output gain) required to produce the desired level L _P at the listener distance _DP can be determined by calculating the doubling of one meter to the listener position: OneMeterDoublings=log2(D ₁ ). Next, the listener level at 1 m from the loudspeaker can be calculated: L _1m =L ₁ -OneMeterDoublings*α _dd . Finally, the loudspeaker drive level or DSP output gain can be identified by G _dsp,out =L _1m -L _sens,spkr -L _dsp,FSout -G _amp -G _attnout +L _{sens,spkr,volts} .

在图5的示例中，房间的一端有一个扩音器，这是为了计算在距扩音器11.92米的位置处产生期望的SPLæ°´å¹³(例如72.0dBSPL)所需的DSP输出增益。该SPL水平是宽带并且未加权，因此使用未加权的全范围啁啾测试信号。房间内碰巧有两个麦克风，但是它们与扩音器的距离是未知的，并且扩音器也是未知的。已知的系统参数是：L_dspfsout＝+20.98dBu，G_dsp,out＝-20.27dB(用于啁啾测量的DSP输出增益)，G_amp＝29.64dB，G_attn,out＝-19.1dB，以及L_{sens,spkr,volts}＝+11.25dBu(2.83V)。该程序被概括为七个操作，1)生成啁啾并且测量在两个或更多个位置处的响应。生成单个啁啾并且记录来自两个麦克风的响应。啁啾测量揭示了以下数据：在距扩音器1.89m处，L₁＝82.0dB_SPL，在距扩音器7.23m处，L₂＝73.8dB_SPL，2)计算距离每增加一倍时的房间衰减，α_dd＝-(82.0dB-73.8dB)/log2(7.23m/1.89m)＝-4.24dB/doubling，3)通过首先找到最靠近的麦克风的相对于1m距离的两倍来计算距扬声器1米处的啁啾水平，OneMeterDoublings＝log2(1.89m)＝0.918doublings，现在使用L_1m＝82.0dB_SPL-(0.918doublings)*(-4.24dB/doublings)＝85.9dB_SPL计算1m处的啁啾水平，4)计算扩音器的灵敏度，L_sens,spkr＝85.9dB_SPL-20.98dBu-(-20.27dB)-29.64dB-(-19.1dB)+11.25dBu＝85.9dB_SPL，5)计算从1米处到收听者距离DP的两倍，OneMeterDoublings＝log2(11.92m)＝3.575doublings，6)使用L_1m＝72dB_SPL-(3.575doublings)*(-4.236dB/doubling)＝87.15dB_SPL计算距扩音器1米处所需的水平。最后，计算产生该水平所需的DSP输出增益，G_dsp,out＝87.15dB_SPL-85.9dB_SPL-20.98dBu-29.64dB-(-19.1dB)+11.25dBu＝-19.01dB。在该示例中，使用-20.27dB的DSP输出增益在离扩音器11.92米处啁啾被测量为72.0dB_SPL，因此在该示例中计算的输出增益与实际增益相差(20.27-19.01)＝1.26dB。In the example of FIG. 5 , there is a loudspeaker at one end of the room, and this is to calculate the DSP output gain required to produce the desired SPL level (e.g., 72.0 dB SPL) at a location 11.92 meters from the loudspeaker. This SPL level is broadband and unweighted, so an unweighted full-range chirp test signal is used. There happen to be two microphones in the room, but their distances from the loudspeaker are unknown, and the loudspeaker is also unknown. The known system parameters are: L _dspfsout = +20.98 dBu, G _dsp,out = -20.27 dB (DSP output gain for chirp measurement), G _amp = 29.64 dB, G _attn,out = -19.1 dB, and L _{sens,spkr,volts} = +11.25 dBu (2.83 V). The procedure is summarized in seven operations, 1) Generate chirp and measure the response at two or more locations. A single chirp is generated and the responses from both microphones are recorded. The chirp measurement revealed the following data: at 1.89m from the loudspeaker, _L1 = 82.0dB _SPL , at 7.23m from the loudspeaker, _L2 = 73.8dB _SPL , 2) Calculate the room attenuation for each doubling of the distance, _αdd = -(82.0dB-73.8dB)/log2(7.23m/1.89m) = -4.24dB/doubling, 3) Calculate the chirp level at 1 meter from the loudspeaker by first finding twice the distance relative to 1m for the closest microphone, OneMeterDoublings = log2(1.89m) = 0.918doublings, now using _L1m = 82.0dB _SPL -(0.918doublings)*(-4.24dB/doublings) = 85.9dB _SPL calculates the chirp level at 1m, 4) calculates the sensitivity of the loudspeaker, L _sens,spkr = 85.9dB _SPL -20.98dBu-(-20.27dB)-29.64dB-(-19.1dB)+11.25dBu=85.9dB _SPL , 5) calculates twice the distance DP from 1 meter to the listener, OneMeterDoublings=log2(11.92m)=3.575doublings, 6) calculates the required level at 1 meter from the loudspeaker using L _1m = 72dB _SPL -(3.575doublings)*(-4.236dB/doubling)=87.15dB _SPL . Finally, the DSP output gain required to produce this level is calculated, G _dsp,out = 87.15dB _SPL - 85.9dB _{SPL -} 20.98dBu - 29.64dB - (-19.1dB) + 11.25dBu = -19.01dB. In this example, the chirp is measured as 72.0dB _SPL at 11.92 meters from the loudspeaker using a DSP output gain of -20.27dB, so the calculated output gain in this example differs from the actual gain by (20.27-19.01) = 1.26dB.

该程序基于距未知的扩音器1.89m和7.23m处测量的单个啁啾，计算出规定的DSP输出增益为-19.0dB，以在距扩音器11.9米处实现72.0dB_SPL的SPL水平，并且该计算的增益基于位于两个麦克风的范围之外的11.9m处的实际测量水平误差是1.26dB。如果有限的DSP资源仅允许按顺序一次测量在一个麦克风处的水平，则必须以不同的方式计算水平差(L1-L2)。如果对于每个麦克风，增加测试信号直到达到期望的SPL水平，然后记录所需的SPL水平和输出增益，则dB水平差是：dB_diff＝(L1-G_dBout1)-(L2-G_dBout2)。当麦克风1比麦克风2更靠近扬声器时，该dB_diff将是正值。通常L1和L2将是相同的，但是更近的麦克风将需要更低的输出增益以实现两个麦克风相同的SPL水平，因此Gd_Bout1将更低，从而使dB_diff为正值。The program calculates a prescribed DSP output gain of -19.0 dB to achieve an SPL level of 72.0 dB _SPL at 11.9 meters from the loudspeaker based on single chirps measured at 1.89 m and 7.23 m from the unknown loudspeaker, and the calculated gain is 1.26 dB error based on the actual measured level at 11.9 m, which is out of range of both microphones. If limited DSP resources allow only one microphone at a time to be measured in sequence, the level difference (L1-L2) must be calculated differently. If for each microphone, the test signal is increased until the desired SPL level is reached, and then the desired SPL level and output gain are recorded, the dB level difference is: dB _diff = (L1-G _dBout1 )-(L2-G _dBout2 ). This dB _diff will be positive when microphone 1 is closer to the loudspeaker than microphone 2. Normally L1 and L2 will be the same, but the closer microphone will need a lower output gain to achieve the same SPL level for both microphones, so Gd _Bout1 will be lower, making dB _diff a positive value.

在另一示例中，建立输入麦克风增益水平可以包括：如果麦克风具有已知的输入灵敏度，则可以针对最佳动态范围设置包括模拟前置放大器增益的DSP输入增益。例如，如果在房间中麦克风位置处期望的最大声压级是100dB SPL，则可以将增益设置为100dBSPL，并且这将提供满刻度值。如果输入增益被设置得太高，则在前置放大器或A/D转换器中可能发生削波。如果输入增益被设置得太低，则将产生弱信号和过多噪声(因自动增益控制(AGC)而失真)。In another example, establishing the input microphone gain level may include: If the microphone has a known input sensitivity, the DSP input gain including the analog preamplifier gain may be set for the best dynamic range. For example, if the maximum sound pressure level expected at the microphone location in the room is 100dB SPL, the gain may be set to 100dB SPL and this will provide a full scale value. If the input gain is set too high, clipping may occur in the preamplifier or A/D converter. If the input gain is set too low, a weak signal and excessive noise (distortion due to automatic gain control (AGC)) will result.

如果麦克风不具有已知的输入灵敏度，则可以使用来自最靠近每个麦克风输入的扩音器的啁啾响应信号水平和时间差(TOF)信息来估计麦克风灵敏度。如果麦克风不具有全向拾取模式，以及由于麦克风的未知频率响应引起的其他影响，则该估计将具有来自扩音器的未知离轴衰减和/或麦克风的未知离轴衰减的误差。If the microphones do not have known input sensitivities, the chirp response signal levels and time difference (TOF) information from the loudspeakers closest to each microphone input can be used to estimate the microphone sensitivity. This estimate will have errors from the unknown off-axis attenuation of the loudspeakers and/or the unknown off-axis attenuation of the microphones if the microphones do not have omnidirectional pickup patterns, as well as other effects due to the unknown frequency response of the microphones.

在确定扩音器均衡时，理想地，每个扩音器将被均衡，以补偿其频率响应不规则性以及附近表面对低频的增强。如果麦克风的频率响应是已知的，则可以在减去麦克风的已知响应之后经由啁啾去卷积来测量每个扩音器响应。此外，如果扩音器具有已知的频率响应，则只有房间的响应可以被确定。其原因是因为房间中的表面反射可以在所测量的响应中引起梳状滤波，这是不期望的。梳状滤波是一种时域现象，不能用频域滤波来校正。必须考虑对脉冲响应中的表面反射的检测，以便如果可以检测到在时间上更久远的主要反射，则可以对它们进行脉冲响应的开窗口(windowed－out)，从而将它们从用于导出DSP滤波器的频率响应中去除。When determining loudspeaker equalization, ideally each loudspeaker would be equalized to compensate for its frequency response irregularities and the enhancement of low frequencies by nearby surfaces. If the frequency response of the microphones is known, each loudspeaker response can be measured via chirp deconvolution after subtracting the known response of the microphones. Furthermore, if the loudspeakers have known frequency responses, only the response of the room can be determined. The reason for this is because surface reflections in the room can cause comb filtering in the measured response, which is undesirable. Comb filtering is a time domain phenomenon and cannot be corrected with frequency domain filtering. The detection of surface reflections in the impulse response must be taken into account so that if major reflections further back in time can be detected, they can be windowed-out of the impulse response, thereby removing them from the frequency response used to derive the DSP filter.

如果麦克风的频率响应是未知的，则频率响应测量不能区分由扩音器引起的不规则性和由麦克风引起的不规则性。如果做出未知麦克风和扩音器的频率响应并且将所有校正应用于扩音器输出路径，则麦克风中的缺陷将针对扩音器被过度校正，并且在来自远侧扬声器的音频呈现期间为房间的远侧的收听者提供差的声音。类似地，如果将所有校正应用于麦克风输入路径，则扩音器中的缺陷将针对麦克风被过度校正，并且将为位于近侧扬声器的远端的收听者产生差的声音。“分割差值(splitting the difference)”并且将一半校正应用于麦克风输入并且将一半应用于扩音器输出是不可行的策略，并且不太可能产生良好的声音。If the frequency response of the microphone is unknown, the frequency response measurement cannot distinguish between irregularities caused by the loudspeaker and irregularities caused by the microphone. If the frequency response of the unknown microphone and loudspeaker is made and all corrections are applied to the loudspeaker output path, then imperfections in the microphone will be overcorrected for the loudspeaker and provide poor sound for the listener on the far side of the room during the audio presentation from the far side speaker. Similarly, if all corrections are applied to the microphone input path, imperfections in the loudspeaker will be overcorrected for the microphone and will produce poor sound for the listener located far from the near side speaker. "Splitting the difference" and applying half of the correction to the microphone input and half to the loudspeaker output is an unworkable strategy and is unlikely to produce good sound.

将使用标准无限冲激响应(IIR)参数式滤波器进行均衡。有限冲激响应(FIR)滤波器将不太适用于本申请，因为它们具有线性而不是对数或倍频带频率分辨率，这可能需要非常多的分接头用于低频滤波器，并且当(一个户多个)确切收听位置未知时，有限冲激响应(FIR)滤波器将不太适用。通过“反滤波”来确定IIR滤波器，使得所测量的幅度响应的倒数被用作目标以“最佳适配”级联的参数式滤波器。实际限制在于自动均衡滤波器将校正响应的程度(dB)和距离/宽度/窄度(Hz)。已知由反滤波进行的来自脉冲响应的频率响应校正对于源和收听者位置是精确的。因为麦克风位置是唯一已知的值，为了使每个扩音器在所有收听位置都有良好的声音，所以将对频率响应执行整体求平均，使得在某个倍频带平滑被应用之后，由扩音器拾取的来自所有麦克风的响应将被一起求平均。该程序对安装者是透明的，因为可以使用单个扩音器啁啾同时记录来自所有麦克风的响应。Standard infinite impulse response (IIR) parametric filters will be used for equalization. Finite impulse response (FIR) filters will not be suitable for this application because they have linear rather than logarithmic or octave band frequency resolution, which may require a very large number of taps for low frequency filters, and when the exact listening position (one or more) is unknown, the finite impulse response (FIR) filter will not be suitable. The IIR filter is determined by "inverse filtering" so that the inverse of the measured amplitude response is used as a target to "best fit" the cascaded parametric filter. The actual limitation is the degree (dB) and distance/width/narrowness (Hz) of the response that the automatic equalization filter will correct. It is known that the frequency response correction from the impulse response performed by inverse filtering is accurate for the source and listener position. Because the microphone position is the only known value, in order to make each loudspeaker have a good sound at all listening positions, the frequency response will be averaged overall so that after a certain octave band smoothing is applied, the response from all microphones picked up by the loudspeaker will be averaged together. The procedure is transparent to the installer, as a single loudspeaker chirp can be used to simultaneously record responses from all microphones.

一个示例可以包括麦克风均衡程序，当麦克风频率响应未知时，对未知扩音器的均衡是不实际的并且不应该被尝试，因此未知麦克风的频率响应不能被确定。然而，如果扩音器频率响应是已知的，则对未知麦克风的麦克风均衡是可能的。经由啁啾去卷积的麦克风均衡处理将利用存储在固件中的扩音器的已知响应，该已知响应将被减去以得到麦克风的响应。应该针对每个扩音器重复该过程，使得可以将整体求平均应用于测量的频率响应。将通过扩音器均衡中描述的反滤波方法来确定每个麦克风的均衡器设置。One example may include a microphone equalization routine, where equalization for an unknown microphone is not practical and should not be attempted when the microphone frequency response is unknown, and thus the frequency response of the unknown microphone cannot be determined. However, if the microphone frequency response is known, microphone equalization for an unknown microphone is possible. The microphone equalization process via chirp deconvolution will utilize the known response of the microphone stored in the firmware, which will be subtracted to obtain the response of the microphone. This process should be repeated for each microphone so that overall averaging can be applied to the measured frequency response. The equalizer settings for each microphone will be determined by the inverse filtering method described in microphone equalization.

一旦扩音器和麦克风水平被设置并且频率响应不规则性被均衡，则可以基于房间的RT60测量来设置扬声器值和水平。可以通过计算脉冲的Schroeder逆积分来获得混响时间(RT60)，并且RT60是声音在具有扩散声场的空间中衰减60dB所花费的时间的测量，这意味着房间足够大，使得来自源的反射以相同的水平响应能量从所有方向到达麦克风。一旦(一个或多个)RT60值是已知的，则可以设置NLP水平，其中，当混响尾部比AEC的有效尾部长度长时，使用通常更积极的NLP设置。Once the loudspeaker and microphone levels are set and the frequency response irregularities are equalized, the speaker values and levels can be set based on the RT60 measurement of the room. The reverberation time (RT60) can be obtained by calculating the Schroeder inverse integral of the impulse, and RT60 is a measure of the time it takes for a sound to decay by 60 dB in a space with a diffuse sound field, meaning that the room is large enough that reflections from the source respond with the same level of energy reaching the microphone from all directions. Once the RT60 value(s) are known, the NLP level can be set, with a generally more aggressive NLP setting being used when the reverberation tail is longer than the effective tail length of the AEC.

另一示例可以包括设置输出限制器。如果功率放大器增益是已知的并且扩音器功率评级是已知的，则可以设置DSP输出限制器以保护扩音器。另外，如果扩音器灵敏度是已知的，则限制器可以进一步减小最大信号水平，以保护收听者免受过大的声级的影响。对于大多数管理员来说，保持功率增益/灵敏度的增益值信息和类似记录不是可行的选择。此外，即使增益值是已知的，但是扬声器是被错误接线/错误配置的，例如桥接接线不正确的情况，则增益将是不正确的并且导致不正确的功率限制设置。因此，SPL限制是更被期望的操作。Another example may include setting an output limiter. If the power amplifier gain is known and the loudspeaker power rating is known, a DSP output limiter may be set to protect the loudspeaker. Additionally, if the loudspeaker sensitivity is known, the limiter may further reduce the maximum signal level to protect the listener from excessive sound levels. For most administrators, maintaining gain value information and similar records of power gain/sensitivity is not a viable option. Furthermore, even if the gain value is known, but the loudspeaker is miswired/misconfigured, such as in the case of incorrect bridge wiring, the gain will be incorrect and result in an incorrect power limit setting. Therefore, SPL limiting is a more desirable operation.

根据附加的示例实施例，测量会议室的语音清晰度评级(SIR)可以包括测量房间中一个语音源到一个收听者位置的语音传输指数(STI)。替代地，也可以检查多个语音源(例如，天花板扬声器)和房间周围的多个收听位置以识别最佳STI和对应的SIR。此外，会议情形中的语音源可以是远程的，其中远程麦克风、远程房间和传输声道都可能影响收听者的语音清晰度体验。在通常将同时使用多个扩音器的会议室中，应该在同时播放所有“语音会议”扬声器的情况下测量STI。语音会议扬声器指示在会议期间通常会打开的所有扬声器，而专用于音乐回放的所有扬声器将被关闭。原因是收听者通常同时收听来自所有语音会议扬声器的语音，因此语音清晰度将受到所有扬声器的影响，因此评级应在所有语音会议扬声器都处于激活状态时进行测量。与单个扩音器相比，在所有语音会议扩音器都开启的情况下测量的STI可能更好，也可能更差，这取决于背景噪声水平、房间中的回声和混响、扬声器之间的间距等。According to an additional example embodiment, measuring the speech intelligibility rating (SIR) of a conference room may include measuring a speech transmission index (STI) from a speech source in the room to a listener position. Alternatively, multiple speech sources (e.g., ceiling speakers) and multiple listening positions around the room may also be checked to identify the best STI and corresponding SIR. In addition, the speech source in a conference situation may be remote, where the remote microphone, remote room, and transmission channel may all affect the listener's speech intelligibility experience. In a conference room where multiple loudspeakers will typically be used simultaneously, the STI should be measured with all "voice conference" loudspeakers playing simultaneously. The voice conference loudspeakers indicate all loudspeakers that are typically turned on during a meeting, while all loudspeakers dedicated to music playback will be turned off. The reason is that the listener typically listens to the speech from all voice conference loudspeakers at the same time, so the speech intelligibility will be affected by all loudspeakers, so the rating should be measured when all voice conference loudspeakers are activated. Compared to a single loudspeaker, the STI measured with all voice conference loudspeakers turned on may be better or worse, depending on the background noise level, echo and reverberation in the room, the spacing between the loudspeakers, etc.

自动调谐过程可以使用来自会议系统的麦克风而不使用附加的测量麦克风，因此所获得的STI测量值可以是放置在收听者的确切耳朵位置处的测量麦克风的真实STI值的代表。因为会议室具有若干收听者位置，并且可以具有若干会议麦克风，所以将通过在所有‘N’个麦克风处同时执行测量，计算‘N’个STI值，然后对这些值求平均以给出单个房间单个STI值来获得最佳STI评级。这将是在所有会议麦克风位置处所测量的平均STI值，该平均STI值是在所有收听者位置处的平均STI值的代表。自动调谐程序被设计成一次一个地顺序通过每个输出扬声器区域并且同时测量所有麦克风。实时STI分析器任务是DSP密集型的，并且一次只能测量单个麦克风输入。因此，这对测量‘N’个麦克风处的STI值并且求平均值提出了实际限制。对于最精确的STI值，应该同时播放所有的语音会议扬声器。因此，在自动调谐过程中可能需要某些策略来测量多个麦克风处的STI。The automatic tuning process can use the microphone from the conference system without using an additional measurement microphone, so the STI measurement value obtained can be a representative of the true STI value of the measurement microphone placed at the exact ear position of the listener. Because the conference room has several listener positions and can have several conference microphones, the best STI rating will be obtained by performing measurements at all 'N' microphones simultaneously, calculating 'N' STI values, and then averaging these values to give a single STI value for a single room. This will be the average STI value measured at all conference microphone positions, which is a representative of the average STI value at all listener positions. The automatic tuning program is designed to pass through each output speaker zone one at a time and measure all microphones simultaneously. The real-time STI analyzer task is DSP-intensive and can only measure a single microphone input at a time. Therefore, this places practical limitations on measuring the STI values at 'N' microphones and averaging. For the most accurate STI value, all voice conference speakers should be played simultaneously. Therefore, some strategies may be needed to measure the STI at multiple microphones during the automatic tuning process.

一种策略可以包括虽然所有扬声器都播放STI信号，但是仅在第一扬声器迭代期间测量STI，并且使用第一麦克风进行测量。另一种方式是使用被确定为位于中间位置的麦克风进行测量，该中间位置由在IR的计算中所测量的扬声器到麦克风的距离确定。另一种方式是，对于每个扬声器区域迭代，测量在下一个麦麦克风输入上的STI，使得可以对多个STI测量值求平均。该方式具有缺点，例如如果只有一个扬声器区域，则只有第一麦克风被测量。如果扬声器区域比麦克风少，则这可能错过位于中间的麦克风，并且操作该方式的时间最长。One strategy may include measuring the STI only during the first speaker iteration, with the first microphone being used for the measurement, although all speakers play the STI signal. Another approach is to use a microphone determined to be in a middle position for the measurement, where the middle position is determined by the speaker-to-microphone distance measured in the calculation of the IR. Another approach is to measure the STI on the next microphone input for each speaker zone iteration, so that multiple STI measurements can be averaged. This approach has disadvantages, for example if there is only one speaker zone, only the first microphone is measured. If there are fewer speaker zones than microphones, this may miss the microphone in the middle, and this approach takes the longest time to operate.

还应当注意，STI值通常被理解为表示该房间中的语音传输质量。对于远程会议系统，收听者体验的语音传输质量具有三个分量：扩音器和他/她所坐的房间的STI、电子传输声道的STI以及远端麦克风和房间的STI。因此，通过自动调谐程序计算的STI值仅仅是构成收听者的语音清晰度体验的三个分量中的一个分量的代表。然而，这样的信息仍然是有用的，因为可以获得用户或安装者可以控制的近端组件的评分。例如，用户/安装者可以使用自动调谐STI评分来评估使用两种不同声学处理设计对STI的相对改进。It should also be noted that the STI value is generally understood to represent the quality of speech transmission in the room. For teleconferencing systems, the quality of speech transmission experienced by a listener has three components: the STI of the loudspeaker and the room in which he/she sits, the STI of the electronic transmission soundtrack, and the STI of the far-end microphone and room. Therefore, the STI value calculated by the automatic tuning program is only a representative of one of the three components that make up the listener's speech intelligibility experience. However, such information is still useful because scores for near-end components that the user or installer can control can be obtained. For example, a user/installer can use the automatic tuning STI score to evaluate the relative improvement in STI using two different acoustic treatment designs.

自动均衡算法能够自动地将任何房间中的任何扩音器的频率响应均衡到可以由平直线和/或参数式曲线定义的任何期望的响应形状。该算法未被设计为在活动程序音频事件期间实时运行，而是被设计为在系统设置程序期间实时运行。该算法仅考虑并且均衡对数幅度频率响应(分贝对频率)，而不试图均衡相位。该算法基本上设计了一组最佳滤波器，这些滤波器的频率响应与所测量的响应的倒数非常匹配，以便使其变平或使其重塑为某个其他期望的响应。该算法仅使用钟形(升压或切断参数式滤波器)、低通或高通类型的单-双二阶IIR滤波器。可以使用FIR滤波器，但是选择IIR滤波器是因为它们的计算效率、更好的低频分辨率，并且更适合于在房间中的宽广的收听区域进行空间平均或均衡。The automatic equalization algorithm is able to automatically equalize the frequency response of any loudspeaker in any room to any desired response shape that can be defined by a flat line and/or a parametric curve. The algorithm is not designed to run in real time during an active program audio event, but is designed to run in real time during a system setup procedure. The algorithm only considers and equalizes the logarithmic magnitude frequency response (decibels versus frequency) without attempting to equalize the phase. The algorithm basically designs a set of optimal filters whose frequency responses closely match the inverse of the measured response in order to flatten it or reshape it to some other desired response. The algorithm uses only single-biquad IIR filters of the bell-shaped (boosted or cut-off parametric filters), low-pass or high-pass type. FIR filters can be used, but IIR filters are selected because of their computational efficiency, better low-frequency resolution, and are more suitable for spatial averaging or equalization over a wide listening area in a room.

当执行均衡过程时，首先识别期望的目标频率响应。通常，这将是具有低频滚降和高频滚降的平坦响应，以避免该过程设计将尝试实现来自限频扩音器的不可实现结果的滤波器组。目标中频带响应不必是平坦的，并且该过程允许以双二阶滤波器阵列形式的任何任意目标频率响应。该过程还允许用户对要应用的总DSP滤波器组设置最大dB升压或切断极限值。When performing the equalization process, the desired target frequency response is first identified. Typically, this will be a flat response with low frequency roll-off and high frequency roll-off to avoid the process designing a filter bank that will try to achieve an unachievable result from a limited frequency loudspeaker. The target mid-band response does not have to be flat, and the process allows for any arbitrary target frequency response in the form of a biquad filter array. The process also allows the user to set a maximum dB boost or cutoff limit on the total DSP filter bank to be applied.

图6A示出了用于为音频系统执行自动调谐程序的过程。参考图6A，该过程可以包括识别由控制器控制的网络上的多个分离的扬声器612，向第一扬声器提供第一测试信号并且向第二扬声器提供第二测试信号614，检测由控制器控制的一个或多个麦克风处的第一测试信号和第二测试信号，以及基于对不同测试信号的分析自动建立扬声器调谐输出参数616。调谐参数可以被应用于数字DSP参数集，该数字DSP参数集被应用于音频环境中的各种扬声器和麦克风。FIG6A illustrates a process for performing an automatic tuning procedure for an audio system. Referring to FIG6A , the process may include identifying a plurality of separate speakers on a network controlled by a controller 612, providing a first test signal to a first speaker and providing a second test signal to a second speaker 614, detecting the first test signal and the second test signal at one or more microphones controlled by the controller, and automatically establishing speaker tuning output parameters based on analysis of the different test signals 616. The tuning parameters may be applied to a digital DSP parameter set that is applied to various speakers and microphones in an audio environment.

第一测试信号可以与第二测试信号的频率不同。可以在第一时间提供第一测试信号，并且可以在比第一时间晚的第二时间提供第二测试信号。该过程还可以包括通过经由一个或多个麦克风测量环境噪声水平，基于对不同测试信号的分析自动建立扬声器调谐输出参数，基于第一测试信号和第二测试信号确定脉冲响应，以及基于脉冲响应和环境噪声水平确定第一扬声器和第二扬声器使用的扬声器输出水平。该过程还可以包括基于第一扬声器和第二扬声器的输出确定频率响应，对与第一测试信号和第二测试信号相关联的值求平均以获得一个或多个麦克风的平均声压级(SPL)、距所有一个或多个麦克风的平均距离以及从一个或多个麦克风测量的平均频率响应中的一个或多个。该过程还可以包括启动作为针对第一扬声器和第二扬声器中的每个扬声器继续的迭代程序的验证程序。该过程还可以包括执行自动均衡程序，以识别第一扬声器和第二扬声器对所期望的响应形状的频率响应，并且识别具有与所测量的频率响应的倒数非常匹配的频率响应的一个或多个最佳滤波器。The first test signal may be at a different frequency than the second test signal. The first test signal may be provided at a first time, and the second test signal may be provided at a second time later than the first time. The process may also include automatically establishing speaker tuning output parameters based on analysis of different test signals by measuring ambient noise levels via one or more microphones, determining impulse responses based on the first test signal and the second test signal, and determining speaker output levels used by the first speaker and the second speaker based on the impulse responses and the ambient noise level. The process may also include determining a frequency response based on the output of the first speaker and the second speaker, averaging the values associated with the first test signal and the second test signal to obtain one or more of an average sound pressure level (SPL) of one or more microphones, an average distance from all one or more microphones, and an average frequency response measured from one or more microphones. The process may also include initiating a verification procedure that is an iterative procedure that continues for each of the first speaker and the second speaker. The process may also include executing an automatic equalization procedure to identify the frequency response of the first speaker and the second speaker to a desired response shape, and identifying one or more optimal filters having a frequency response that closely matches the inverse of the measured frequency response.

图6B示出了用于为音频系统执行自动调谐程序的过程。参考图6B，该过程可以包括在特定房间环境中识别由控制器控制的网络上的多个扬声器和一个或多个麦克风652，提供测试信号以从每个放大器声道和多个扬声器顺序播放654，同时监视来自一个或多个麦克风的测试信号以检测操作扬声器和放大器声道656，向多个扬声器提供附加测试信号以确定调谐参数658，检测由控制器控制的一个或多个麦克风处的附加测试信号662，以及基于检测到的附加测试信号自动建立房间环境的背景噪声水平和噪声频谱664。FIG6B illustrates a process for performing an automatic tuning procedure for an audio system. Referring to FIG6B , the process may include identifying a plurality of speakers and one or more microphones on a network controlled by a controller in a particular room environment 652, providing a test signal to be played sequentially from each amplifier channel and the plurality of speakers 654, while monitoring the test signal from the one or more microphones to detect operating speakers and amplifier channels 656, providing additional test signals to the plurality of speakers to determine tuning parameters 658, detecting additional test signals at the one or more microphones controlled by the controller 662, and automatically establishing a background noise level and noise spectrum for the room environment based on the detected additional test signals 664.

该过程还可以包括同时监视来自一个或多个麦克风的测试信号以识别是否有任何放大器输出声道未被连接到多个扬声器。附加测试信号可以包括在第一时间提供的第一测试信号和在比第一时间晚的第二时间提供的第二测试信号。该过程还可以包括自动建立多个扬声器中的每个扬声器的频率响应，以及每个放大器声道和对应扬声器的灵敏度水平。灵敏度水平基于特定房间环境的目标声压级(SPL)。该过程还可以包括识别从一个或多个麦克风中的每个麦克风到多个扬声器中的每个扬声器的距离、特定房间环境的房间混响时间、用于实现目标SPL的每个扬声器的声道水平设置、用于归一化每个扬声器的频率响应和实现目标房间频率响应的每个扬声器的声道均衡设置、对于特定房间环境最佳的声学回声消除参数、对降低由麦克风检测到的针对特定房间环境的背景噪声最佳的降噪参数以及当针对房间环境未检测到声音时对降低环境噪声最佳的非线性处理参数。该过程还可以包括启动作为针对多个扬声器中的每个扬声器继续的迭代程序的验证程序，并且该验证程序包括再次检测由控制器控制的一个或多个麦克风处的附加测试信号以验证目标SPL和目标房间频率响应。The process may also include monitoring test signals from one or more microphones simultaneously to identify whether any amplifier output channel is not connected to a plurality of speakers. The additional test signals may include a first test signal provided at a first time and a second test signal provided at a second time later than the first time. The process may also include automatically establishing a frequency response of each speaker in a plurality of speakers, and a sensitivity level of each amplifier channel and a corresponding speaker. The sensitivity level is based on a target sound pressure level (SPL) of a specific room environment. The process may also include identifying a distance from each microphone in the one or more microphones to each speaker in the plurality of speakers, a room reverberation time of a specific room environment, a channel level setting for each speaker to achieve a target SPL, a channel equalization setting for each speaker to normalize the frequency response of each speaker and achieve a target room frequency response, an acoustic echo cancellation parameter that is optimal for a specific room environment, a noise reduction parameter that is optimal for reducing background noise detected by a microphone for a specific room environment, and a nonlinear processing parameter that is optimal for reducing ambient noise when no sound is detected for a room environment. The process may also include initiating a verification procedure that continues as an iterative procedure for each of the plurality of speakers and includes again detecting additional test signals at one or more microphones controlled by the controller to verify the target SPL and target room frequency response.

图7示出了用于执行自动音频系统设置配置的示例过程。参考图7，该过程可以包括识别连接到由控制器控制的网络的多个扬声器和麦克风712，向用于施加测试信号的多个扬声器分配初步输出增益714，测量从麦克风检测到的环境噪声716，同时记录来自所有麦克风的啁啾响应718，对所有啁啾响应去卷积以确定对应数量的脉冲响应722，以及测量每个麦克风的平均声压级(SPL)以基于SPL的平均值获得SPL水平724。An example process for performing automatic audio system setup configuration is shown in FIG7. Referring to FIG7, the process may include identifying a plurality of speakers and microphones connected to a network controlled by a controller 712, assigning preliminary output gains 714 to the plurality of speakers for applying a test signal, measuring ambient noise detected from the microphones 716, simultaneously recording chirp responses from all microphones 718, deconvolving all chirp responses to determine a corresponding number of impulse responses 722, and measuring an average sound pressure level (SPL) of each microphone to obtain an SPL level 724 based on an average of the SPLs.

测量从麦克风检测到的环境噪声可以包括检查过多噪声。对于每个麦克风输入信号，该过程可以包括识别主脉冲峰值，以及识别从多个扬声器中的一个或多个扬声器到每个麦克风的距离。该过程可以包括确定每个麦克风输入信号的频率响应，以及基于频率响应将补偿值施加到每个麦克风。该过程还可以包括对频率响应求平均以获得空间平均响应，以及执行对空间平均响应的自动均衡以匹配目标响应值。该过程还可以包括基于SPL水平以及最近麦克风和最远麦克风的距离确定与房间相关联的衰减值，以及基于SPL水平和衰减值确定提供所有麦克风的平均距离处的目标声级的输出增益。Measuring the ambient noise detected from the microphone may include checking for excessive noise. For each microphone input signal, the process may include identifying a main pulse peak, and identifying a distance from one or more of the plurality of speakers to each microphone. The process may include determining a frequency response for each microphone input signal, and applying a compensation value to each microphone based on the frequency response. The process may also include averaging the frequency responses to obtain a spatially averaged response, and performing automatic equalization of the spatially averaged response to match a target response value. The process may also include determining an attenuation value associated with the room based on an SPL level and the distances of the nearest microphone and the farthest microphone, and determining an output gain that provides a target sound level at an average distance of all microphones based on the SPL level and the attenuation value.

图8示出了用于对音频系统执行自动均衡程序的示例过程。参考图8，该过程可以包括确定对从一个或多个扬声器检测到的所测量的啁啾信号的频率响应812，基于高极限值和低极限值确定频率响应的平均值814，从目标响应减去所测量的响应，其中，目标响应基于一个或多个滤波器频率816，基于该减法结果确定具有可听参数的限频目标滤波器818，以及基于由限频目标滤波器定义的区域来应用无限冲激响应(IIR)双二阶滤波器以均衡一个或多个扬声器的频率响应822。FIG8 illustrates an example process for performing an automatic equalization procedure on an audio system. Referring to FIG8 , the process may include determining a frequency response to a measured chirp signal detected from one or more speakers 812, determining an average of the frequency response based on an upper limit and a lower limit 814, subtracting the measured response from a target response, wherein the target response is based on one or more filter frequencies 816, determining a frequency-limited target filter having audible parameters based on the subtraction result 818, and applying an infinite impulse response (IIR) biquad filter based on a region defined by the frequency-limited target filter to equalize the frequency response of the one or more speakers 822.

平均值被设置为零分贝，并且目标响应基于与一个或多个双二阶滤波器相关联的一个或多个频率。基于目标响应确定目标滤波器可以包括确定目标零交叉和目标滤波器导数零点。该过程还可以包括基于检测到的幅度峰值来限制目标滤波器的分贝，以创建被限制的滤波器，以及将该被限制的滤波器添加到滤波器组。该过程还可以包括将未被限制的均衡滤波器添加到所测量的响应以提供未被限制的校正响应。该过程还可以包括从目标响应减去未被限制的校正响应以提供新的目标滤波器。The average value is set to zero decibels, and the target response is based on one or more frequencies associated with one or more biquad filters. Determining the target filter based on the target response may include determining a target zero crossing and a target filter derivative zero point. The process may also include limiting the decibels of the target filter based on the detected amplitude peak to create a limited filter, and adding the limited filter to the filter bank. The process may also include adding an unlimited equalization filter to the measured response to provide an unlimited correction response. The process may also include subtracting the unlimited correction response from the target response to provide a new target filter.

图9示出了用于确定一个或多个增益值以应用于音频系统的示例过程。参考图9，该过程可以包括对扬声器施加一组初始功率和增益参数912，经由扬声器914播放刺激信号，测量所播放的刺激的频率响应信号916，确定麦克风位置处的声级和距一个或多个扬声器的预定距离处的声级918，基于麦克风位置处的声级和距扬声器的预定距离处的声级之差来确定麦克风位置处的增益922，以及将增益施加到扬声器输出924。FIG9 shows an example process for determining one or more gain values to apply to an audio system. Referring to FIG9 , the process may include applying a set of initial power and gain parameters to a speaker 912, playing a stimulus signal via a speaker 914, measuring a frequency response signal of the played stimulus 916, determining a sound level at a microphone location and a sound level at a predetermined distance from one or more speakers 918, determining a gain at a microphone location based on a difference between the sound level at the microphone location and the sound level at a predetermined distance from the speaker 922, and applying the gain to a speaker output 924.

预定距离可以是与用户相对于扬声器位置可能所处的位置相关联的设定距离，例如一米。该过程还可以包括检测距扬声器第一距离的麦克风处和距扬声器第二距离(比第一距离远)的第二麦克风处的刺激信号，并且在两个麦克风处同时执行该检测。该过程还可以包括确定第一距离处的第一声压级和第二距离处的第二声压级。该过程还可以包括基于第一声压级和第二声压级之差确定扬声器的衰减。该过程还可以包括：当扬声器由参考电压驱动时，基于在距扬声器预定距离处测量的声压级来确定扬声器的灵敏度。The predetermined distance may be a set distance associated with a position where a user may be relative to the speaker position, such as one meter. The process may also include detecting a stimulus signal at a microphone at a first distance from the speaker and at a second microphone at a second distance from the speaker (longer than the first distance), and performing the detection at both microphones simultaneously. The process may also include determining a first sound pressure level at the first distance and a second sound pressure level at the second distance. The process may also include determining the attenuation of the speaker based on the difference between the first sound pressure level and the second sound pressure level. The process may also include determining the sensitivity of the speaker based on the sound pressure level measured at a predetermined distance from the speaker when the speaker is driven by a reference voltage.

图10示出了用于识别语音清晰度评级或语音传输指数的过程。参考图10，该过程可以包括启动自动调谐程序1012，经由一个或多个麦克风检测与两个或更多个位置处的多个扬声器的输出相关联的声音测量1014，确定与麦克风的数量相等的语音传输索引(STI)值的数量1016，以及对语音传输索引值求平均以识别单个语音传输索引值1018。FIG10 illustrates a process for identifying a speech intelligibility rating or speech transmission index. Referring to FIG10, the process may include initiating an automatic tuning procedure 1012, detecting sound measurements associated with the output of a plurality of speakers at two or more locations via one or more microphones 1014, determining a number of speech transmission index (STI) values equal to the number of microphones 1016, and averaging the speech transmission index values to identify a single speech transmission index value 1018.

该过程还可以包括在多个扬声器同时提供输出信号的同时测量STI值的数量。在多个扬声器同时提供输出信号的同时测量STI值的数量可以包括使用一个麦克风。在多个扬声器同时提供输出信号的同时测量STI值的数量可以包括使用多个麦克风中的一个麦克风，并且该一个麦克风被识别为最靠近多个扬声器的位置中的中间位置。对语音传输索引值求平均以识别单个语音传输索引值可以包括测量‘N’个麦克风处的STI值，其中‘N’大于1，以及对‘N’个值求平均以识别特定环境的单个STI值。The process may also include measuring the number of STI values while the multiple speakers are providing output signals simultaneously. Measuring the number of STI values while the multiple speakers are providing output signals simultaneously may include using one microphone. Measuring the number of STI values while the multiple speakers are providing output signals simultaneously may include using one microphone among the multiple microphones, and the one microphone is identified as a middle position among the positions closest to the multiple speakers. Averaging the voice transmission index values to identify a single voice transmission index value may include measuring STI values at ‘N’ microphones, where ‘N’ is greater than 1, and averaging the ‘N’ values to identify a single STI value for a specific environment.

自动调谐可以仅使用会议系统通常所需的组件而不使用其他仪器来自动地测量会议音频系统和对应房间的语音清晰度。可以与第三方功率放大器和扩音器一起使用自动调谐。因为这些组件的增益和灵敏度是未知的，所以自动调谐处理使用唯一的宽带多音逐步增加信号，连同经由声学时延自动测量并且使用声速计算的扬声器到麦克风的距离，来快速地确定这些参数，直到其达到在麦克风处已知的SPL水平。使用该技术，自动调谐可以确定对应组件的增益和灵敏度，以及来自扩音器的SPL水平。快速逐步增加宽带多音信号，并且为系统参数的自动确定提供优化。基于各种滤波器，自动调谐自动均衡算法快速均衡多个扬声器区域。此外，附加增强被添加到该算法中。Automatic tuning can automatically measure the speech intelligibility of conference audio systems and corresponding rooms using only the components commonly required for conference systems without other instruments. Automatic tuning can be used with third-party power amplifiers and loudspeakers. Because the gain and sensitivity of these components are unknown, the automatic tuning process uses a unique broadband multi-tone step-up signal, together with the distance from the speaker to the microphone automatically measured via acoustic delay and calculated using the speed of sound, to quickly determine these parameters until it reaches the known SPL level at the microphone. Using this technology, automatic tuning can determine the gain and sensitivity of the corresponding component, as well as the SPL level from the loudspeaker. Rapidly increase the broadband multi-tone signal step by step, and provide optimization for the automatic determination of system parameters. Based on various filters, the automatic tuning automatic equalization algorithm quickly equalizes multiple speaker areas. In addition, additional enhancements are added to the algorithm.

该过程可以包括根据水平和增益来分析电声系统，以确定实现期望的声学等级所需的增益，以及优化用于最大动态范围的增益结构。历史上以“dB SPL”表示声压级。通常用单位“dB”表示声级，意味着它实际上是相对于0dB＝20u帕斯卡的绝对级。现代国际标准将声压级表示为Lp/(20uPa)或缩写为Lp。然而，Lp通常也用于表示声级的变量，而不是声级的单位。为了避免任何混淆，在该分析中，声压级将总是被表示为“dBa”，表示绝对学声等级，并且与过时的“dB SPL”相同。“dBa”不应该与“dBA”混淆，dBA通常是针对A加权声级表示的单位。在该分析中，‘L’总是作为绝对量的水平变量，而‘G’总是作为相对量的增益变量。因为等式包含具有不同单位(电学对声学)的变量，但是仍然以分贝表示，所以为了清楚起见，在{}中明确地示出了单位。The process can include analyzing the electroacoustic system in terms of level and gain to determine the gain required to achieve the desired acoustic level, and optimizing the gain structure for maximum dynamic range. Sound pressure level has historically been expressed in "dB SPL". Sound level is usually expressed in the unit "dB", meaning that it is actually an absolute level relative to 0dB = 20u Pascal. Modern international standards express sound pressure level as Lp/(20uPa) or abbreviated as Lp. However, Lp is also commonly used to represent the variable of sound level rather than the unit of sound level. To avoid any confusion, in this analysis, sound pressure level will always be expressed as "dBa", which means absolute acoustic level and is the same as the outdated "dB SPL". "dBa" should not be confused with "dBA", which is usually a unit expressed for A-weighted sound level. In this analysis, 'L' is always used as a level variable in absolute terms, and 'G' is always used as a gain variable in relative terms. Because the equation contains variables with different units (electrical vs. acoustic), but are still expressed in decibels, the units are explicitly shown in {} for clarity.

该分析被分成两个明显不同的信号路径，从声源(讲话者218)到DSP内部处理的输入路径，以及从DSP内部处理到从扩音器输出的声学等级的路径。这两条路径则各自具有两种变化。输入信号路径具有模拟对数字麦克风变化，而输出路径具有模拟对数字功率放大器变化(就其输入信号而非其功率放大技术而言是数字的)。为了一致性和简单起见，所有信号衰减被表示为将具有负值的增益。例如，GP-S＝LP-LSpkr是从扩音器(1米处)到人员的增益，并且该值可能为例如-6dB。这些增益在图中被示出为笔直的箭头，但是实际上声音路径由表面反射和来自房间周围的扩散声组成。显然，房间的脉冲响应将揭示房间行为的细节，但是在该分析中，我们仅关注例如由粉红噪声引起的非时间稳态声级。为了简化该分析，这些多个声音路径被归并到增益为‘G’的单个路径中。通过测量GP-S和GM-P，可以识别收听者位置处的已知声级，以及设定的DSP输出增益和输入前置放大器增益。因为在收听者位置处没有测量麦克风，所以GP-S和GM-P是被估计的。然而，我们可以精确地测量GM-S，并且基于典型的会议室声学“经验法则”对GP-S和GM-P进行估计。为了一致性和简单起见，所有信号衰减被表示为将具有负值的增益。例如，GP-S＝LP-LSpkr是从扩音器(1米处)到人员的增益，并且该值可能是例如-6dB。这些增益在图中被示出为笔直的箭头，但是实际上声音路径由表面反射和来自房间周围的扩散声组成。显然，房间的脉冲响应将揭示房间行为的细节，但是在该分析中，识别例如由粉红噪声引起的非时间稳态声级。为了简化该分析，多个声音路径被归并到增益为G的单个路径中。GP-S和GM-P被测量，所以可以识别收听者位置处的已知声级，以及最佳地设置DSP输出增益和输入前置放大器增益。The analysis is divided into two distinct signal paths, the input path from the sound source (talker 218) to the DSP internal processing, and the path from the DSP internal processing to the acoustic level output from the loudspeaker. Each of these two paths then has two variations. The input signal path has an analog to digital microphone variation, while the output path has an analog to digital power amplifier variation (digital in terms of its input signal rather than its power amplification technology). For consistency and simplicity, all signal attenuations are represented as gains that will have negative values. For example, GP-S=LP-LSpkr is the gain from the loudspeaker (at 1 meter) to the person, and this value may be, for example, -6dB. These gains are shown as straight arrows in the figure, but in reality the sound path consists of surface reflections and diffuse sound from around the room. Obviously, the impulse response of the room will reveal the details of the room behavior, but in this analysis, we are only concerned with non-time-steady sound levels caused, for example, by pink noise. In order to simplify the analysis, these multiple sound paths are merged into a single path with a gain of 'G'. By measuring GP-S and GM-P, the known sound level at the listener position can be identified, as well as the set DSP output gain and input preamplifier gain. Because there is no measurement microphone at the listener position, GP-S and GM-P are estimated. However, we can accurately measure GM-S and estimate GP-S and GM-P based on typical conference room acoustic "rules of thumb". For consistency and simplicity, all signal attenuations are expressed as gains that will have negative values. For example, GP-S=LP-LSpkr is the gain from the loudspeaker (at 1 meter) to the person, and the value may be, for example, -6dB. These gains are shown as straight arrows in the figure, but in reality the sound path consists of surface reflections and diffuse sound from around the room. Obviously, the impulse response of the room will reveal the details of the room behavior, but in this analysis, non-time-steady sound levels caused by, for example, pink noise are identified. To simplify the analysis, multiple sound paths are merged into a single path with a gain of G. GP-S and GM-P are measured so a known sound level at the listener's position can be identified and the DSP output gain and input preamplifier gain optimally set.

自动调谐可以仅使用会议系统通常所需的组件而不使用其他仪器来自动地测量会议音频系统和对应房间的语音清晰度。可以与第三方功率放大器和扩音器一起使用自动调谐。因为这些组件的增益和灵敏度是未知的，所以自动调谐处理使用唯一的宽带多音逐步增加信号，连同经由声学时延自动测量并且使用声速计算的扬声器到麦克风的距离，来快速地确定这些参数，直到其达到在麦克风处已知的SPL水平。使用该技术，自动调谐可以确定对应组件的增益和灵敏度，以及来自扩音器的SPL水平。快速逐步增加宽带多音信号，并且为系统参数的自动确定提供优化。基于各种滤波器，自动调谐自动均衡算法快速均衡多个扬声器区域。此外，附加增强被添加到该算法中。Automatic tuning can automatically measure the speech intelligibility of conference audio systems and corresponding rooms using only the components commonly required for conference systems without other instruments. Automatic tuning can be used with third-party power amplifiers and loudspeakers. Because the gain and sensitivity of these components are unknown, the automatic tuning process uses a unique broadband multi-tone step-up signal, together with the distance from the speaker to the microphone automatically measured via acoustic delay and calculated using the speed of sound, to quickly determine these parameters until it reaches the known SPL level at the microphone. Using this technology, automatic tuning can determine the gain and sensitivity of the corresponding component, as well as the SPL level from the loudspeaker. Rapidly increase the broadband multi-tone signal step by step, and provide optimization for the automatic determination of system parameters. Based on various filters, the automatic tuning automatic equalization algorithm quickly equalizes multiple speaker areas. In addition, additional enhancements are added to the algorithm.

一个示例实施例可以包括测量语音清晰度以合理地获得会议室的语音清晰度评级。应该相对于多个语音源(例如天花板扬声器)和房间周围的多个收听位置来识别语音传输指数(STI)。此外，会议情形中的语音源可以是远程的，其中远程麦克风、远程房间和传输声道都可能影响收听者的语音清晰度体验。在具有通常将被同时使用的多个扩音器的会议室中，在逻辑上应该用同时播放的所有“语音会议”扬声器来测量STI。语音会议扬声器表示在会议期间通常会打开的所有扬声器，而专用于音乐回放的所有扬声器将被关闭。原因是收听者通常同时收听来自所有语音会议扬声器的语音，因此语音清晰度将受到所有扬声器的影响，因此评级应在所有语音会议扬声器都处于激活状态时进行测量。与单个扩音器相比，在所有语音会议扩音器都开启的情况下测量的STI可能更好，也可能更差，这取决于背景噪声水平、房间中的回声和混响、扬声器之间的间距等。An example embodiment may include measuring speech intelligibility to reasonably obtain a speech intelligibility rating for a conference room. A speech transmission index (STI) should be identified relative to multiple speech sources (e.g., ceiling speakers) and multiple listening positions around the room. In addition, the speech source in a conference situation may be remote, where the remote microphone, remote room, and transmission channel may all affect the listener's speech intelligibility experience. In a conference room with multiple loudspeakers that are typically used simultaneously, the STI should logically be measured with all "voice conference" speakers that are playing simultaneously. Voice conference speakers represent all speakers that are typically turned on during a meeting, while all speakers dedicated to music playback will be turned off. The reason is that listeners typically listen to speech from all voice conference speakers at the same time, so speech intelligibility will be affected by all speakers, so the rating should be measured when all voice conference speakers are activated. Compared to a single loudspeaker, the STI measured when all voice conference loudspeakers are turned on may be better or worse, depending on the background noise level, echo and reverberation in the room, the spacing between speakers, etc.

因为自动调谐必须使用来自会议系统的麦克风而不是附加的测量麦克风，所以应当注意，来自自动调谐的STI测量值是放置在收听者的耳朵位置处的测量麦克风的真实STI值的代表。因为会议室具有若干收听者位置，并且可以具有若干会议麦克风，所以将通过在所有N个麦克风处同时进行测量，计算N个STI值，然后对这些值求平均以给出单个房间STI值来获得最好的STI评级。这将是在所有会议麦克风位置处测量的平均STI值，该平均STI值又将是所有收听者位置处的平均STI值的代表。(一个或多个)自动调谐算法被设计成一次一个地顺序通过每个输出扬声器区域，并且同时测量所有麦克风。此外，实时STI分析器任务是非常的DSP密集型的，并且一次只能测量单个麦克风输入。因此，这对测量在‘N’个麦克风处STI值并且对这些值求平均提出了实际限制。对于最精确的STI值，应该同时播放所有的语音会议扬声器。Because the automatic tuning must use the microphone from the conference system instead of the additional measurement microphone, it should be noted that the STI measurement from the automatic tuning is a representative of the true STI value of the measurement microphone placed at the listener's ear position. Because the conference room has several listener positions and may have several conference microphones, the best STI rating will be obtained by measuring at all N microphones simultaneously, calculating N STI values, and then averaging these values to give a single room STI value. This will be the average STI value measured at all conference microphone positions, which will in turn be a representative of the average STI value at all listener positions. (One or more) automatic tuning algorithms are designed to pass through each output speaker zone one at a time and measure all microphones simultaneously. In addition, the real-time STI analyzer task is very DSP intensive and can only measure a single microphone input at a time. Therefore, this places practical limitations on measuring STI values at 'N' microphones and averaging these values. For the most accurate STI value, all voice conference speakers should be played simultaneously.

在自动调谐程序中可能用于测量多个麦克风处的STI的几种策略可以包括：作为第一方式，仅在第一扬声器迭代期间测量STI，但是所有扬声器将播放STIPA，然后使用第一麦克风执行测量，但是使用麦克风的测量被确定为位于由在CalcIR状态中测量的扬声器到麦克风的距离所确定的中间位置。另一种方式可以包括：对于每个扬声器区域迭代，测量下一个麦克风输入上的STI，使得可以对多个STI测量值求平均。然而，某些问题可能是如果仅有一个扬声器区域，则将仅测量第一麦克风。如果扬声器区域比麦克风少，则可能错过位于中间的麦克风，并且运行该方式的时间最长。Several strategies that may be used in an auto-tune routine to measure STI at multiple microphones may include: as a first approach, measuring STI only during the first speaker iteration, but all speakers will play the STI, and then performing a measurement using the first microphone, but the measurement using the microphone is determined to be located in the middle position determined by the speaker-to-microphone distance measured in the CalcIR state. Another approach may include: for each speaker zone iteration, measuring the STI on the next microphone input so that multiple STI measurements can be averaged. However, some problems may be that if there is only one speaker zone, only the first microphone will be measured. If there are fewer speaker zones than microphones, the microphone in the middle may be missed and the time to run this approach is the longest.

还应当注意，STI值通常被理解为表示该房间中的语音传输质量。对于远程会议系统，收听者体验的语音传输质量实际上具有三个分量：扩音器和人员所坐的房间的STI、电子传输声道的STI以及远端麦克风和房间的STI。因此，通过自动调谐计算的STI值仅仅是构成收听者语音清晰度体验的三个分量中的一个分量的代表。然而，这仍然可以为用户或安装者可以在事件期间控制的近端组件提供评分。例如，用户/安装者可以使用自动调谐STI评分来评估使用两种不同声学处理设计对STI的相对改进。It should also be noted that the STI value is generally understood to represent the quality of speech transmission in that room. For teleconferencing systems, the quality of speech transmission experienced by the listener actually has three components: the STI of the loudspeaker and the room in which the person is sitting, the STI of the electronic transmission soundtrack, and the STI of the far-end microphone and room. Therefore, the STI value calculated by auto-tune is only a representation of one of the three components that make up the listener's speech intelligibility experience. However, this can still provide a score for near-end components that the user or installer can control during an event. For example, a user/installer can use the auto-tune STI score to evaluate the relative improvement in STI using two different acoustic treatment designs.

自动调谐可以仅使用会议系统通常所需的组件而不使用其他仪器来自动地测量会议音频系统和对应房间的语音清晰度。可以与第三方功率放大器和扩音器一起使用自动调谐。因为这些组件的增益和灵敏度是未知的，所以自动调谐处理使用唯一的宽带多音逐步增加信号，连同经由声学时延自动测量并且使用声速计算的扬声器到麦克风的距离，来快速地确定这些参数，直到其达到在麦克风处已知的SPL水平。使用该技术，自动调谐可以确定对应组件的增益和灵敏度，以及来自扩音器的SPL水平。快速逐步增加宽带多音信号，并且为系统参数的自动确定提供优化。基于各种滤波器，自动调谐自动均衡算法快速均衡多个扬声器区域。此外，附加增强被添加到该算法中。Automatic tuning can automatically measure the speech intelligibility of conference audio systems and corresponding rooms using only the components commonly required for conference systems without other instruments. Automatic tuning can be used with third-party power amplifiers and loudspeakers. Because the gain and sensitivity of these components are unknown, the automatic tuning process uses a unique broadband multi-tone step-up signal, together with the distance from the speaker to the microphone automatically measured via acoustic delay and calculated using the speed of sound, to quickly determine these parameters until it reaches the known SPL level at the microphone. Using this technology, automatic tuning can determine the gain and sensitivity of the corresponding component, as well as the SPL level from the loudspeaker. Rapidly increase the broadband multi-tone signal step by step, and provide optimization for the automatic determination of system parameters. Based on various filters, the automatic tuning automatic equalization algorithm quickly equalizes multiple speaker areas. In addition, additional enhancements are added to the algorithm.

根据一个示例实施例，启动过程序列可以包括基于其在房间中的位置(例如，安装在天花板上，在桌子上等)对已知并且连接到控制器的麦克风进行剖析。此外，用于基于DSP过程产生‘报告卡’或一组测试结果的过程可包括各种测试和检测到的反馈。在一个示例中，启动过程检测与控制器通信的所有设备，例如计算机或类似的计算设备。这些设备可以包括位于房间内的各种麦克风和扬声器。检测过程可以测量房间中设备的表现，调谐扬声器以及调节(一个或多个)扬声器水平。此外，也可以经由数字信号处理技术来确定房间混响(reverb)值和语音清晰度评级。房间混响的麦克风降噪和补偿也可以被确定和设置，用于随后的扬声器和麦克风使用。启动过程可以引起房间评级从第一评级变为第二评级。例如，初始房间评级可以是‘一般的’，并且一旦做出某个扬声器和/或修改，随后的房间评级可以是‘出色的’。此外，图形用户界面可以生成报告或‘报告卡’，该报告或‘报告卡’表示在执行设置/启动过程之前和之后的某些房间特性。报告卡可以被下载作为用于记录目的的文件。各种版本的报告卡可以被生成，并且被显示在与控制器通信的用户设备上或经由控制器设备的显示器显示。如果最终报告卡是‘好’而不是‘非常好’，则可以在报告卡上示出关于如何进一步优化房间音频特性的示例。会议室一般由一起工作的所有设备或大多数音频设备而不仅仅是由独立于其他设备被调谐的一个单独的设备调谐。此外，报告卡可以提供到用于优化房间的音频表现的信息的链接。According to an example embodiment, the startup process sequence may include profiling microphones that are known and connected to the controller based on their location in the room (e.g., mounted on the ceiling, on a table, etc.). In addition, the process for generating a 'report card' or a set of test results based on the DSP process may include various tests and feedback detected. In one example, the startup process detects all devices that communicate with the controller, such as a computer or similar computing device. These devices may include various microphones and speakers located in the room. The detection process may measure the performance of the devices in the room, tune the speakers, and adjust the speaker level (one or more). In addition, room reverberation (reverb) values and speech intelligibility ratings may also be determined via digital signal processing techniques. Microphone noise reduction and compensation for room reverberation may also be determined and set for subsequent speaker and microphone use. The startup process may cause the room rating to change from a first rating to a second rating. For example, the initial room rating may be 'average', and once certain speakers and/or modifications are made, the subsequent room rating may be 'excellent'. In addition, the graphical user interface may generate a report or 'report card' that represents certain room characteristics before and after performing the setup/startup process. The report card may be downloaded as a file for logging purposes. Various versions of the report card may be generated and displayed on a user device in communication with the controller or via a display of the controller device. If the final report card is 'good' rather than 'very good', examples of how to further optimize the room audio characteristics may be shown on the report card. Conference rooms are typically tuned by all or most of the audio devices working together rather than just one single device that is tuned independently of the other devices. In addition, the report card may provide links to information for optimizing the audio performance of the room.

图11示出了自动调谐平台的示例。在一个示例中，可以针对理想的音频特性测试和优化房间或其他类型的音频环境1112。在操作中，当在控制器1128(例如，计算机、用户界面、网络设备)上选择调谐按钮或选项时。启动过程可以通过控制器1128播放音频设置过程开始，该音频设置过程经由描述调谐过程的每个步骤的可听数据文件来指示用户。最初，执行设备检测过程以识别每个扬声器(例如，扬声器1142、1144等)和每个麦克风1132、1134等。交换机1122可以是连接到麦克风1132/1134、扬声器1142/1144和控制器1128的以太网交换机。可以生成识别初始扬声器调谐参数(包括但不限于房间混响、噪声底等)的初始表现测量。在声音序列由扬声器播放并且由麦克风检测到之后，初始表现测量可以表示特定水平的整体质量，例如‘一般’、‘好’、‘非常好’等。可以从扬声器1142/1144中的一个或多个扬声器播放第一音调，然后可以由该扬声器播放在时间、频率、dB水平等方面与第一音调不同的第二音调。麦克风1132/1134可以捕获音频音调并且提供控制器可以处理的信号以识别房间特性并且通过创建包括在报告或其他信息共享工具中的评级或其他指示符来确定目标是否被满足。可以在控制器1128的文件中保存在初始序列期间捕获的信息。每个扬声器可以一次一个地被测试并且由两个麦克风测量，然后下一个扬声器将由两个麦克风测试和测量。扬声器和麦克风的数量可以是任意的，并且针对每种类型的设备可以包括一个、两个或更多个。然后可以通过所计算的DSP参数来修改房间噪声底、混响值和其他值。下一轮测试可以将这些被修改的DSP值应用于扬声器，以确定自初始测试程序以来噪声底、语音清晰度是否已经改善。可以通过播放附加声音并且经由麦克风记录声音来确定最终评级。下一个评级应该比上一个评级更优，并且目的是经由对特定房间中和对于(一个或多个)特定目标或目的的声音测试的多次迭代来达到‘非常好’的评级。FIG. 11 shows an example of an automatic tuning platform. In one example, a room or other type of audio environment 1112 can be tested and optimized for ideal audio characteristics. In operation, when a tuning button or option is selected on a controller 1128 (e.g., a computer, a user interface, a network device). The startup process can start by playing an audio setup process by the controller 1128, which instructs the user via an audible data file describing each step of the tuning process. Initially, a device detection process is performed to identify each speaker (e.g., speakers 1142, 1144, etc.) and each microphone 1132, 1134, etc. The switch 1122 can be an Ethernet switch connected to microphones 1132/1134, speakers 1142/1144, and controller 1128. An initial performance measurement identifying initial speaker tuning parameters (including but not limited to room reverberation, noise floor, etc.) can be generated. After the sound sequence is played by the speaker and detected by the microphone, the initial performance measurement can represent the overall quality of a specific level, such as 'general', 'good', 'very good', etc. A first tone may be played from one or more of the speakers 1142/1144, and then a second tone different from the first tone in terms of time, frequency, dB level, etc. may be played by the speaker. Microphones 1132/1134 may capture audio tones and provide signals that the controller may process to identify room characteristics and determine whether the goals are met by creating ratings or other indicators included in reports or other information sharing tools. The information captured during the initial sequence may be saved in a file of the controller 1128. Each speaker may be tested and measured by two microphones one at a time, and then the next speaker will be tested and measured by two microphones. The number of speakers and microphones may be arbitrary, and may include one, two, or more for each type of device. The room noise floor, reverberation values, and other values may then be modified by the calculated DSP parameters. The next round of testing may apply these modified DSP values to the speakers to determine whether the noise floor, speech clarity, has improved since the initial test procedure. The final rating may be determined by playing additional sounds and recording the sounds via the microphones. The next rating should be better than the previous one, and the aim is to achieve a 'very good' rating through multiple iterations of sound testing in a specific room and for (one or more) specific goals or objectives.

图12示出了根据示例实施例的具有针对特定区域的动态音频分布配置的自动调谐平台配置。参考图12，音频配置包括在特定区域中的扬声器1142/144和麦克风1132/1134。扬声器和麦克风的数量在特定区域中可以不同。位于音频环境中的人员的估计数量可以不同。在一个示例中，可以调整和优化由扬声器1142/144产生的音频，以为目标组或人员数量1152(不占据整个区域)或较大的人员数量1154(占据区域的较大部分)产生特定的音频输出。可以测量房间混响水平和/或语音清晰度，并且可以优化扬声器的表现以基于出席者的预期数量和他们在区域内的位置来适应混响和语音清晰度区域。位于该区域的第一部分区域1152内的人员的第一示例可能需要针对该区域的房间混响水平和语音清晰度和/或其他音频特性的第一优化水平。位于该区域的较大区域1154内的人员的第二示例可能需要针对‘区域’(例如会议厅、会议室、办公室空间等)的房间混响水平和语音清晰度和/或其他音频特性的第二优化水平。Figure 12 shows an automatic tuning platform configuration with a dynamic audio distribution configuration for a specific area according to an example embodiment. With reference to Figure 12, the audio configuration includes a speaker 1142/144 and a microphone 1132/1134 in a specific area. The number of speakers and microphones can be different in a specific area. The estimated number of personnel located in the audio environment can be different. In one example, the audio produced by the speaker 1142/144 can be adjusted and optimized to produce a specific audio output for a target group or number of personnel 1152 (not occupying the entire area) or a larger number of personnel 1154 (occupying a larger portion of the area). The room reverberation level and/or speech clarity can be measured, and the performance of the speaker can be optimized to adapt to the reverberation and speech clarity area based on the expected number of attendees and their positions in the area. The first example of the personnel located in the first partial area 1152 of the area may need a first optimization level for the room reverberation level and speech clarity and/or other audio characteristics of the area. A second example of persons located within a larger area 1154 of the area may require a second optimization level for room reverberation level and speech intelligibility and/or other audio characteristics for the ‘area’ (e.g., a conference hall, meeting room, office space, etc.).

在一个示例中，该区域中的预期人员的数量和/或他们在该区域内的位置可以是输入到音频配置设置过程中的参数或者是例如由检测何时以及多少人员进入和离开特定区域的传感器或其他反馈设备基于房间容量的被识别的变化而动态调整的值。随着出席水平被量化，音频输出可以被修改并且调整以产生具有不同混响和/或语音清晰度输出值的音频输出，这取决于扬声器的数量和它们在该区域内的位置。例如，如果一个或两个扬声器位于该区域的前部或该区域的第一半部分中，则当优化这些前部区域扬声器的扬声器输出时，尤其是当预期的出席人未被预期占据该区域的最远部分时，整个区域的混响值可能不太重要。In one example, the number of expected persons in the zone and/or their location within the zone may be parameters input into the audio configuration setup process or values that are dynamically adjusted based on identified changes in room capacity, such as by sensors or other feedback devices that detect when and how many persons enter and leave a particular zone. As attendance levels are quantified, the audio output may be modified and adjusted to produce audio outputs having different reverberation and/or speech intelligibility output values, depending on the number of speakers and their location within the zone. For example, if one or two speakers are located at the front of the zone or in the first half of the zone, the reverberation value for the entire zone may be less important when optimizing the speaker output of these front zone speakers, especially when the expected attendees are not expected to occupy the farthest portion of the zone.

图13示出了根据示例实施例的在音频设置程序期间与控制器通信的计算设备的示例用户界面。参考图13，两个示例用户界面示出了在对扬声器系统进行优化之后的初始启动周期1310和优化启动周期1320。可以根据特定的评级级别来测量和分析各种判据。例如，基于由扬声器输出识别的和由麦克风测量的测量信号，房间简档最初可以被识别为具有中等调谐水平、中等混响水平和中等房间噪声水平。所识别的测量水平可以提供需要进行的相对调节量以优化各种测量水平。一旦扬声器输出缺陷被识别，就可以根据用于优化的各种判据所需的修改量来计算扬声器调整。这样的值可以包括语音传输指数、语音清晰度值、数字滤波器值、房间混响值、噪声调整值等。所得到的优化启动周期可以是更高的等级，例如与‘好’的初始值相比较的‘非常好’。该值与特定的指数或数值相关联，该特定的指数或数值与扬声器输出测量值相关联。FIG. 13 illustrates an example user interface of a computing device communicating with a controller during an audio setup procedure according to an example embodiment. Referring to FIG. 13 , two example user interfaces illustrate an initial startup period 1310 and an optimized startup period 1320 after optimizing a speaker system. Various criteria may be measured and analyzed according to specific rating levels. For example, based on measurement signals identified by speaker output and measured by a microphone, a room profile may initially be identified as having a medium tuning level, a medium reverberation level, and a medium room noise level. The identified measurement levels may provide the relative amount of adjustment that needs to be made to optimize the various measurement levels. Once a speaker output defect is identified, speaker adjustments may be calculated based on the amount of modification required for the various criteria for optimization. Such values may include a speech transmission index, a speech intelligibility value, a digital filter value, a room reverberation value, a noise adjustment value, and the like. The resulting optimized startup period may be a higher rating, such as ‘very good’ compared to an initial value of ‘good’. The value is associated with a specific index or value that is associated with the speaker output measurement value.

图14示出了根据示例实施例的房间噪声表现测量的示例表。参考图14，表1420表示与dBA噪声底的特定数值、阈值和/或数值范围匹配的一些评级。低噪声底(例如小于30dBA)可以被认为是非常好的。其他值是dBA的范围，并且也可以存在极限值，例如50dBA是噪声底的‘差’评级的基准。超过50dBA的任何值都可以被认为作为房间噪声的标准是不可接受的。FIG. 14 shows an example table of room noise performance measurements according to an example embodiment. Referring to FIG. 14 , table 1420 represents some ratings that match specific values, thresholds, and/or ranges of values for the dBA noise floor. A low noise floor (e.g., less than 30 dBA) can be considered very good. Other values are ranges of dBA, and there can also be extreme values, e.g., 50 dBA is the benchmark for a ‘poor’ rating of the noise floor. Any value over 50 dBA can be considered unacceptable as a standard for room noise.

图15示出了根据示例实施例的语音清晰度测量的示例。参考图15，量表1520表示用于语音传输指数(STI)和通用清晰度标度(CIS)的一组刻度值。阈值和范围表示报告值的配对，例如‘不好’、‘差’、‘一般’、‘好’和‘非常好’。测量值可以被识别并且与结果输出的刻度值进行比较。用于展示初始音频房间评级和优化音频房间评级的用户界面的一个示例可示出在初始扬声器调谐程序之后测量房间音频的预发射过程是‘好的’，该初始扬声器调谐过程包括从扬声器播放声音并且经由麦克风记录声音以确定房间的各种音频参数和特性。FIG15 illustrates an example of speech intelligibility measurement according to an example embodiment. Referring to FIG15 , a scale 1520 represents a set of scale values for a speech transmission index (STI) and a common intelligibility scale (CIS). Thresholds and ranges represent pairs of reported values, such as ‘bad’, ‘poor’, ‘fair’, ‘good’, and ‘very good’. The measured values may be identified and compared to the scale values of the resulting output. An example of a user interface for displaying an initial audio room rating and an optimized audio room rating may show that a pre-transmission process for measuring room audio is ‘good’ after an initial speaker tuning procedure that includes playing sound from a speaker and recording sound via a microphone to determine various audio parameters and characteristics of the room.

根据示例实施例，使用另一示例使用界面来表示房间噪声表现和语音清晰度的评级值。第一示例表明，基于以分贝(dBA)为单位的特定噪声底水平，房间噪声表现可以是‘差’、‘一般’、‘好’、‘优’和‘非常好’。语音清晰度评级也可以被确定为在0和1之间的语音传输指数(STI)。音频调节的类型可以包括应用于处于特定水平(例如‘中等’水平)的一个或多个扬声器的降噪、在‘中等’水平处应用的回声降低、可用声道的数量(例如两个)、所使用的声道的数量(例如两个)等。麦克风也可以连同降噪水平的类型、回声降低水平等被识别。According to an example embodiment, another example usage interface is used to represent rating values for room noise performance and speech intelligibility. The first example shows that based on a specific noise floor level in decibels (dBA), the room noise performance can be 'poor', 'average', 'good', 'excellent', and 'very good'. The speech intelligibility rating can also be determined as a speech transmission index (STI) between 0 and 1. The type of audio adjustment can include noise reduction applied to one or more speakers at a specific level (e.g., a 'medium' level), echo reduction applied at a 'medium' level, the number of available channels (e.g., two), the number of channels used (e.g., two), etc. Microphones can also be identified along with the type of noise reduction level, the echo reduction level, etc.

还可以识别某些房间特性，例如房间混响的‘混响’(RT60)值，其表征在房间中声音保持可听的时长。高‘混响’时间可能导致会议系统清晰度降低。混响测量还用于调谐麦克风并且将最佳音频质量递递给远端参与人。混响时间与会议室表现相关。例如，房间表现设置混响时间(RT60)对于小于300ms可以是‘非常好’，对于300-400ms可以是‘优’，对于400-500ms可以是‘好’，对于500-1000ms可以是‘中等’，对于大于1000ms可以是‘差’。房间混响(RT60)平均值在445(ms)被认为‘好’。还可以识别每个倍频带的房间混响(RT60)。混响时间取决于音频信号的频率。RT60可以在倍频带上被制图，并且与所推荐的表现图表上的信息重叠。Certain room characteristics may also be identified, such as the 'reverberation' (RT60) value of the room reverberation, which characterizes how long sounds remain audible in the room. A high 'reverberation' time may result in reduced intelligibility in the conferencing system. Reverberation measurements are also used to tune microphones and deliver the best audio quality to far-end participants. Reverberation time is related to conference room performance. For example, the room performance setting reverberation time (RT60) may be 'very good' for less than 300ms, 'excellent' for 300-400ms, 'good' for 400-500ms, 'moderate' for 500-1000ms, and 'poor' for greater than 1000ms. The room reverberation (RT60) average is considered 'good' at 445 (ms). The room reverberation (RT60) for each octave band may also be identified. The reverberation time depends on the frequency of the audio signal. The RT60 may be plotted over the octave bands and overlaid with the information on the recommended performance chart.

启动优化过程可以包括基于房间的所测量的RT60表现对音频系统进行以下调整的启动。此外，可以确定例如在‘低’值处的回声消除非线性处理(NLP)。在该过程的麦克风均衡阶段期间，房间噪声可以包括会议室中干扰语音的任何声音。一般，房间中的噪声越多，理解某人讲话就越困难。噪声源通常包括HVAC通风口、投影仪、照明装置和来自相邻房间的声音。启动过程执行对房间中的噪声水平的测量，然后将适当的降噪水平应用于麦克风。结果是递送到会议呼叫远端的声音聚焦音频信号。The startup optimization process may include a startup that makes the following adjustments to the audio system based on the measured RT60 performance of the room. In addition, the echo cancellation nonlinear processing (NLP) at a 'low' value, for example, may be determined. During the microphone equalization phase of the process, room noise may include any sound in the conference room that interferes with speech. Generally, the more noise there is in the room, the more difficult it is to understand someone speaking. Noise sources typically include HVAC vents, projectors, lighting fixtures, and sounds from adjacent rooms. The startup process performs a measurement of the noise level in the room, and then applies the appropriate noise reduction level to the microphone. The result is a sound-focused audio signal delivered to the far end of the conference call.

平均混响时间与会议室表现相关。房间噪声的水平可以基于频率而变化。噪声标准(NC)曲线可以用于将房间噪声的全频谱示出为单个值。通过识别未被所测量的值触摸的最低NC曲线来找到NC值。会议室的所推荐的NC评级在NC-25和NC-35之间。The average reverberation time is related to the conference room performance. The level of room noise can vary based on frequency. The Noise Criteria (NC) curve can be used to show the full spectrum of room noise as a single value. The NC value is found by identifying the lowest NC curve that is not touched by the measured value. The recommended NC rating for conference rooms is between NC-25 and NC-35.

启动过程可以基于所测量的房间的房间噪声对音频系统进行各种调整。例如，预启动噪声水平平均值可以被识别为‘38dB’SPL A加权，从而应用的降噪水平：‘中等’，而启动优化传输噪声平均值：21dB SPL A加权，从而麦克风声道2可以被确定。可以对这些值进行加权以调整噪声水平。对于扬声器或‘扩音器调谐’过程，每个房间具有将直接影响扬声器表现的声学特征。必须将扬声器调谐到特定房间以确保远端音频是清晰的并且房间用户不会经历听力疲劳。启动过程测量扬声器频率响应并且将该测量值与已知的表现标准进行比较。然后，启动过程自动补偿来自目标响应的变化以确保特定房间内的峰值表现。The startup process can make various adjustments to the audio system based on the measured room noise of the room. For example, the pre-startup noise level average can be identified as '38dB' SPL A-weighted, thereby applying the noise reduction level: 'Medium', while the startup optimized transmission noise average: 21dB SPL A-weighted, thereby microphone channel 2 can be determined. These values can be weighted to adjust the noise level. For the speaker or 'loudspeaker tuning' process, each room has acoustic characteristics that will directly affect the speaker performance. The speakers must be tuned to a specific room to ensure that the far-end audio is clear and that the room users do not experience listening fatigue. The startup process measures the speaker frequency response and compares this measurement to a known performance standard. The startup process then automatically compensates for variations from the target response to ensure peak performance within the specific room.

启动优化可以包括经由从RT60值、信噪比水平、频率响应、失真、总体设备质量等导出输入的复杂过程来确定清晰度。为了简化语音清晰度的报告，大多数标准组织利用报告单个值的测量技术。该值的最常见的标度是语音传输指数(STI)和通用清晰度标度(CIS)。启动过程通过补偿本地房间内的声学缺陷来影响呈现给远端参与人的音频的清晰度。该过程还通过确保房间扬声器在位于房间中的不同位置时被调谐到目标值来增强对远端音频的本地房间语音清晰度。房间的在启动之后以及在通过该过程优化之后的语音清晰度表现可以在被评级为‘非常好’的值，例如0.76。Startup optimization may include determining intelligibility via a complex process that derives inputs from RT60 values, signal-to-noise levels, frequency response, distortion, overall equipment quality, and the like. To simplify reporting of speech intelligibility, most standards organizations utilize measurement techniques that report a single value. The most common scales for this value are the Speech Transmission Index (STI) and the Common Intelligibility Scale (CIS). The startup process affects the clarity of the audio presented to the far-end participants by compensating for acoustic imperfections within the local room. The process also enhances the local room speech intelligibility to the far-end audio by ensuring that the room speakers are tuned to target values when located at different locations in the room. The speech intelligibility performance of the room after startup and after optimization by this process may be at a value rated as 'very good', such as 0.76.

附加的实施例/示例可以包括基于房间中的人的数量以及人在房间中的位置的测量，和可以根据房间中的人的数量以及人在房间中的位置来改变的测量。此外，更多的人可能进来而其他人可能离开，因此，人们坐着(或站着)的地点可能变空和/或充满。因此，可以执行一种场景，其中，基于预期的出席人数和他们将位于/就坐的最可能的位置对房间进行预调谐，基于进入和/或离开由估计的数量所检测或传感器所检测的房间的人对调谐过程进行实时/近实时更新，该传感器在调谐过程之前识别进入和离的人和/或房间中的人的语音。附加的示例包括检测从天花板麦克风和扬声器发出的声音以及信号，该声音和信号可以用于扬声器定位/校准以及调谐房间。Additional embodiments/examples may include measurements based on the number of people in the room and the positions of the people in the room, and measurements that may change based on the number of people in the room and the positions of the people in the room. In addition, more people may come in and others may leave, and therefore, the places where people sit (or stand) may become empty and/or full. Therefore, a scenario can be performed in which the room is pre-tuned based on the expected number of attendees and the most likely locations where they will be located/seated, and the tuning process is updated in real time/near real time based on people entering and/or leaving the room detected by the estimated number or detected by a sensor that recognizes the voices of people entering and leaving and/or people in the room before the tuning process. Additional examples include detecting sounds and signals emitted from ceiling microphones and speakers, which can be used for speaker positioning/calibration and tuning the room.

根据一个示例实施例，启动过程序列可以包括基于其在房间中的位置(例如，安装在天花板上，在桌子上等)对已知并且连接到控制器的麦克风进行剖析。此外，用于基于DSP过程产生‘报告卡’或一组测试结果的过程可包括各种测试和检测到的反馈。According to an example embodiment, the startup process sequence may include profiling microphones that are known and connected to the controller based on their location in the room (e.g., mounted on the ceiling, on a table, etc.). Additionally, a process for generating a 'report card' or set of test results based on the DSP process may include various tests and detected feedback.

在一个示例中，启动过程检测与控制器通信的所有设备，例如计算机或类似的计算设备。这些设备可以包括位于房间内的各种麦克风和扬声器。检测过程可以测量房间中设备的表现，调谐扬声器以及调节(一个或多个)扬声器水平。此外，也可以经由数字信号处理技术来确定房间混响值和语音清晰度评级。房间混响的麦克风降噪和补偿也可以被确定和设置，用于随后的扬声器和麦克风使用。启动过程可以引起房间评级从第一评级变为第二评级。例如，初始房间评级可以是‘一般’，随后的房间评级可以是‘非常好'。此外，图形用户界面可以生成报告或‘报告卡’，该报告或‘报告卡’表示在执行设置/启动过程之前和之后的某些房间特性。可以下载报告卡。各种版本的报告卡可以被生成，并且被显示在与控制器通信的用户设备上或经由控制器设备的显示器显示。如果最终报告卡是‘好’而不是‘非常好’，则可以在报告卡上示出关于如何进一步优化房间音频特性的示例。会议室由一起工作的所有设备而不仅仅是由独立于其他设备被调谐的一个单独的设备调谐。可以经由web浏览器在线查看报告和/或该报告可以从web或网络源被下载到工作站。In one example, the startup process detects all devices in communication with the controller, such as a computer or similar computing device. These devices may include various microphones and speakers located in the room. The detection process may measure the performance of the devices in the room, tune the speakers, and adjust the speaker level (one or more). In addition, the room reverberation value and speech intelligibility rating may also be determined via digital signal processing techniques. Microphone noise reduction and compensation for room reverberation may also be determined and set for subsequent speaker and microphone use. The startup process may cause the room rating to change from a first rating to a second rating. For example, the initial room rating may be 'average' and the subsequent room rating may be 'very good'. In addition, the graphical user interface may generate a report or 'report card' that represents certain room characteristics before and after performing the setup/startup process. The report card may be downloaded. Various versions of the report card may be generated and displayed on a user device in communication with the controller or via a display of the controller device. If the final report card is 'good' instead of 'very good', examples of how to further optimize the room audio characteristics may be shown on the report card. The conference room is tuned by all devices working together rather than just by a single device that is tuned independently of other devices. The report may be viewed online via a web browser and/or may be downloaded to a workstation from a web or network source.

在一个示例中，当在软件应用接口中手动或虚拟地按压控制器上的调谐按钮时，启动过程可以通过控制器播放音频设置过程开始，该音频设置过程经由提供音频的音频处理数据文件指示用户解释该过程的每个操作。最初，执行设备检测过程以识别每个扬声器(例如，多个扬声器)和每个麦克风等。交换机可以是连接到麦克风、扬声器和控制器的以太网交换机。可以生成识别初始扬声器调谐参数(包括但不限于房间混响、噪声底等)的初始表现测量。在声音序列由扬声器播放并且由麦克风检测到之后，初始表现测量可以表示特定水平的整体质量，例如‘一般’、‘好’、‘非常好’。可以播放第一音调，然后播放在时间、频率、dB水平等方面与第一音调不同的第二音调。可以在控制器的文件中保存在初始序列期间捕获的信息。每个扬声器可以一次一个地被测试并且由两个麦克风测量，然后下一个扬声器将由两个麦克风测试和测量。扬声器和麦克风的数量可以是任意的，并且针对每种类型的设备可以包括一个、两个或更多个。然后可以通过所计算的DSP参数来修改房间噪声底、混响值和其他值。下一轮测试可以将这些被修改的DSP值应用于扬声器，以确定自初始测试程序以来噪声底、语音清晰度是否已经改善。可以通过播放附加声音并且经由麦克风记录声音来确定最终评级。下一个评级应该比上一个评级更优，并且目的是达到‘非常好’的评级。该过程也可以是自主的并且可以不需要用户交互，然而，音频和/或LED可以发射信号以向任何观察者提供对测试过程的更新。此外，可以经由音频信号提供初步和调整/最终表现评级，以通知初始和最终音频状态的任何使用。In one example, when a tuning button on a controller is pressed manually or virtually in a software application interface, the startup process can start by the controller playing an audio setup process that instructs the user to explain each operation of the process via an audio processing data file that provides audio. Initially, a device detection process is performed to identify each speaker (e.g., multiple speakers) and each microphone, etc. The switch can be an Ethernet switch connected to the microphone, the speaker, and the controller. An initial performance measurement that identifies initial speaker tuning parameters (including but not limited to room reverberation, noise floor, etc.) can be generated. After the sound sequence is played by the speaker and detected by the microphone, the initial performance measurement can represent a specific level of overall quality, such as 'average', 'good', 'very good'. A first tone can be played, and then a second tone that is different from the first tone in terms of time, frequency, dB level, etc. can be played. The information captured during the initial sequence can be saved in a file in the controller. Each speaker can be tested and measured by two microphones one at a time, and then the next speaker will be tested and measured by two microphones. The number of speakers and microphones can be arbitrary, and can include one, two or more for each type of device. The room noise floor, reverberation values and other values can then be modified by the calculated DSP parameters. The next round of testing can apply these modified DSP values to the speakers to determine whether the noise floor, speech clarity has improved since the initial test procedure. The final rating can be determined by playing additional sounds and recording the sounds via a microphone. The next rating should be better than the previous rating, and the goal is to achieve a 'very good' rating. The process can also be autonomous and may not require user interaction, however, audio and/or LEDs can emit signals to provide updates to any observer on the test process. In addition, preliminary and adjusted/final performance ratings can be provided via audio signals to notify any use of the initial and final audio status.

基于以分贝(dBA)为单位的特定噪声底水平，房间噪声表现可以被评级为‘差’、‘一般’、‘好’、‘优’和‘非常好’。语音清晰度评级也可以被确定为在0和1之间的语音传输指数(STI)。音频调节的类型可以包括应用于处于特定水平(例如‘中等’水平)的一个或多个扬声器的降噪、在‘中等’水平处应用的回声降低、可用声道的数量(例如两个)、所使用的声道的数量(例如两个)。麦克风也可以连同降噪水平的类型、回声降低水平等被识别。Room noise performance may be rated as 'poor', 'fair', 'good', 'excellent', and 'very good' based on a specific noise floor level in decibels (dBA). Speech intelligibility ratings may also be determined as a speech transmission index (STI) between 0 and 1. Types of audio adjustments may include noise reduction applied to one or more speakers at a specific level (e.g., a 'medium' level), echo reduction applied at a 'medium' level, the number of channels available (e.g., two), the number of channels used (e.g., two). Microphones may also be identified along with the type of noise reduction level, echo reduction level, etc.

还可以识别某些房间特性，例如房间混响(RT60)值，其表征在房间中声音保持可听的时长。高混响时间可能导致会议系统清晰度降低。混响测量还用于调谐麦克风并且将最佳音频质量递递给远端参与人。混响时间与会议室表现相关。例如，房间表现设置混响时间(RT60)对于小于300ms可以是‘非常好’，对于300-400ms可以是‘优’，对于400-500ms可以是好，对于500-1000ms可以是‘中等’，对于大于1000ms可以是‘差’。房间混响(RT60)平均值在445(ms)为‘好’。还可以识别每个倍频带的房间混响(RT60)。混响时间取决于音频信号的频率。RT60可以在倍频带上被制图，并且与所推荐的表现图表上的信息重叠。Certain room characteristics may also be identified, such as the room reverberation (RT60) value, which characterizes how long sounds remain audible in the room. High reverberation time may result in reduced intelligibility in the conferencing system. Reverberation measurements are also used to tune microphones and deliver optimal audio quality to far-end participants. Reverberation time is related to conference room performance. For example, the room performance setting reverberation time (RT60) may be 'very good' for less than 300ms, 'excellent' for 300-400ms, 'good' for 400-500ms, 'medium' for 500-1000ms, and 'poor' for greater than 1000ms. The room reverberation (RT60) average is 'good' at 445 (ms). The room reverberation (RT60) for each octave band may also be identified. The reverberation time depends on the frequency of the audio signal. The RT60 may be plotted over the octave bands and overlaid with the information on the recommended performance chart.

启动优化过程可以包括基于房间的所测量的RT60表现对音频系统进行以下调整的启动。此外，可以确定例如在‘低’值处的回声消除非线性处理(NLP)。在该过程的麦克风均衡阶段期间，房间噪声可以包括会议室中干扰语音的任何声音。一般，房间中的噪声越多，理解某人讲话就越困难。噪声源通常包括HVAC通风口、投影仪、照明装置和来自相邻房间的声音。启动过程执行对房间中的噪声水平的测量，然后将适当的降噪水平应用于麦克风。结果是递送到会议呼叫远端的声音聚焦音频信号。The startup optimization process may include a startup that makes the following adjustments to the audio system based on the measured RT60 performance of the room. In addition, the echo cancellation nonlinear processing (NLP) at a 'low' value, for example, may be determined. During the microphone equalization phase of the process, room noise may include any sound in the conference room that interferes with speech. Generally, the more noise there is in the room, the more difficult it is to understand someone speaking. Noise sources typically include HVAC vents, projectors, lighting fixtures, and sounds from adjacent rooms. The startup process performs a measurement of the noise level in the room, and then applies the appropriate noise reduction level to the microphone. The result is a sound-focused audio signal delivered to the far end of the conference call.

平均混响时间与会议室表现相关。房间噪声的水平可以基于频率而变化。噪声标准(NC)曲线可以用于将房间噪声的全频谱示出为单个值。通过识别未被所测量的值触摸的最低NC曲线来找到NC值。会议室的所推荐的NC评级在NC-25和NC-35之间。The average reverberation time is related to the conference room performance. The level of room noise can vary based on frequency. The Noise Criteria (NC) curve can be used to show the full spectrum of room noise as a single value. The NC value is found by identifying the lowest NC curve that is not touched by the measured value. The recommended NC rating for conference rooms is between NC-25 and NC-35.

启动过程可以基于所测量的房间的房间噪声对音频系统进行各种调整。例如，预启动噪声水平平均值可以被识别为‘38dB’SPL A加权，从而应用的降噪水平：‘中等’，而启动优化传输噪声平均值：21dB SPL A加权，从而麦克风声道2可以被确定。可以对这些值进行加权以调整噪声水平。对于扬声器或‘扩音器调谐’过程，每个房间具有将直接影响扬声器表现的声学特征。必须将扬声器调谐到特定房间以确保远端音频是清晰的并且房间用户不会经历听力疲劳。启动过程测量扬声器频率响应并且将该测量值与已知的表现标准进行比较。然后，启动过程自动补偿来自目标响应的变化以确保特定房间内的峰值表现。The startup process can make various adjustments to the audio system based on the measured room noise of the room. For example, the pre-startup noise level average can be identified as '38dB' SPL A-weighted, thereby applying the noise reduction level: 'Medium', while the startup optimized transmission noise average: 21dB SPL A-weighted, thereby microphone channel 2 can be determined. These values can be weighted to adjust the noise level. For the speaker or 'loudspeaker tuning' process, each room has acoustic characteristics that will directly affect the speaker performance. The speakers must be tuned to a specific room to ensure that the far-end audio is clear and that the room users do not experience listening fatigue. The startup process measures the speaker frequency response and compares this measurement to a known performance standard. The startup process then automatically compensates for variations from the target response to ensure peak performance within the specific room.

启动优化可以包括经由从RT60值、信噪比水平、频率响应、失真、总体设备质量等导出输入的复杂过程来确定清晰度。为了简化语音清晰度的报告，大多数标准组织利用报告单个值的测量技术。该值的最常见的标度是语音传输指数(STI)和通用清晰度标度(CIS)。启动过程通过补偿本地房间内的声学缺陷来影响呈现给远端参与人的音频的清晰度。该过程还通过确保房间扬声器在位于房间中的不同位置时被调谐到目标值来增强对远端音频的本地房间语音清晰度。房间的在启动之后以及在通过该过程优化之后的语音清晰度表现可以在被评级为‘非常好’的值，例如0.76。Startup optimization may include determining intelligibility via a complex process that derives inputs from RT60 values, signal-to-noise levels, frequency response, distortion, overall equipment quality, and the like. To simplify reporting of speech intelligibility, most standards organizations utilize measurement techniques that report a single value. The most common scales for this value are the Speech Transmission Index (STI) and the Common Intelligibility Scale (CIS). The startup process affects the clarity of the audio presented to the far-end participants by compensating for acoustic imperfections within the local room. The process also enhances the local room speech intelligibility to the far-end audio by ensuring that the room speakers are tuned to target values when located at different locations in the room. The speech intelligibility performance of the room after startup and after optimization by this process may be rated at a value of 'very good', such as 0.76.

附加的实施例/示例可以包括基于房间中的人以及人在房间中的位置的测量，和可以根据房间中的人以及人在房间中的位置来改变的测量。此外，更多的人可能进来而其他人可能离开，因此，人们坐着(或站着)的地点可能变空和/或充满。因此，可以执行一种场景，其中，基于预期的出席人数和他们将位于/就坐的最可能的位置对房间进行预调谐，基于进入和/或离开由估计的数量所检测或传感器所检测的房间的人对调谐过程进行实时/近实时更新，该传感器在调谐过程之前识别进入和离的人和/或房间中的人的语音。附加示例包括检测从天花板麦克风和扬声器发出的声音以及信号(绿色和红色)，该声音以及信号可以用于扬声器定位/校准以及调谐房间。Additional embodiments/examples may include measurements based on people in the room and their positions in the room, and measurements that may change based on people in the room and their positions in the room. In addition, more people may come in and others may leave, and therefore, places where people sit (or stand) may become empty and/or full. Therefore, a scenario can be performed in which the room is pre-tuned based on the expected number of attendees and the most likely locations where they will be located/seated, and the tuning process is updated in real time/near real time based on people entering and/or leaving the room detected by the estimated number or detected by a sensor that recognizes the voices of people entering and leaving and/or people in the room before the tuning process. Additional examples include detecting sounds and signals (green and red) emitted from ceiling microphones and speakers, which can be used for speaker positioning/calibration and tuning the room.

图16示出了根据示例实施例的用于确定房间的初始音频简档并且优化音频简档的过程的示例流程图。一个示例过程可以包括经由控制器检测区域中的一个或多个麦克风和一个或多个扬声器1612。该检测可以通过由控制器检测的无线或有线信号来实现，该控制器可以包括网络设备、计算机和/或类似的数据处理设备。该过程还可以包括测量一个或多个麦克风和一个或多个扬声器的音频表现水平，以识别噪声底和混响水平中的一者或多者1614，基于音频表现水平识别初始房间表现评级1616。评级可以是与(一个或多个)所测量的值的特定数值相关联的离散水平。该过程还可以包括将优化的扬声器调谐水平应用于一个或多个扬声器和一个或多个麦克风1618，优化的扬声器调谐水平可以包括幅度、滤波器、电压和修改扬声器表现的其他数字信号。该过程还可以包括经由一个或多个麦克风基于所应用的优化的扬声器调谐水平测量一个或多个扬声器的音频表现水平1620；以及基于所应用的优化的扬声器调谐生成报告以识别优化的房间表现评级1622。可以对优化的扬声器表现进行分级和监视，以确保实现优化水平。FIG. 16 shows an example flow chart of a process for determining an initial audio profile for a room and optimizing the audio profile according to an example embodiment. An example process may include detecting one or more microphones and one or more speakers in an area via a controller 1612. The detection may be achieved via a wireless or wired signal detected by a controller, which may include a network device, a computer, and/or similar data processing device. The process may also include measuring the audio performance level of the one or more microphones and one or more speakers to identify one or more of the noise floor and the reverberation level 1614, and identifying an initial room performance rating 1616 based on the audio performance level. The rating may be a discrete level associated with a specific numerical value of the measured value (one or more). The process may also include applying an optimized speaker tuning level to one or more speakers and one or more microphones 1618, and the optimized speaker tuning level may include amplitude, filter, voltage, and other digital signals that modify the speaker performance. The process may also include measuring the audio performance level of one or more speakers based on the applied optimized speaker tuning level 1620 via one or more microphones; and generating a report based on the applied optimized speaker tuning to identify the optimized room performance rating 1622. Optimized loudspeaker performance can be graded and monitored to ensure that optimized levels are achieved.

该过程还可以包括应用初始扬声器调谐水平以应用于一个或多个扬声器。该过程还可以包括：测量音频表现水平包括基于目标值(例如目标水平或作为理想水平的基准)来测量混响值、噪声水平和语音清晰度值。该报告可以包括基于优化的扬声器调谐水平的房间等级、房间混响补偿和房间噪声水平。初始房间表现评级被分配第一等级，而优化的房间表现评级被分配比第一等级更高和更优的第二等级。该更高的等级可以包括与所测量的值相关联的一个或多个值，所测量的值是不同的并且被认为比初始测量的值更优。一个或多个麦克风和一个或多个扬声器的音频表现水平的测量基于目标水平，并且可以包括识别麦克风的数量、使用中的扬声器的数量和目标声压级。The process may also include applying an initial speaker tuning level to apply to one or more speakers. The process may also include: measuring the audio performance level includes measuring reverberation values, noise levels, and speech intelligibility values based on target values (e.g., target levels or as a benchmark for ideal levels). The report may include a room grade, room reverberation compensation, and room noise level based on the optimized speaker tuning level. The initial room performance rating is assigned a first grade, while the optimized room performance rating is assigned a second grade that is higher and better than the first grade. The higher grade may include one or more values associated with the measured value, which is different and is considered to be better than the initial measured value. The measurement of the audio performance level of one or more microphones and one or more speakers is based on the target level and may include identifying the number of microphones, the number of speakers in use, and the target sound pressure level.

图17示出了根据示例实施例的用于基于理想频率响应来确定房间的初始音频简档并且尝试修改音频简档的过程的示例流程图。参考图17，该过程可以包括经由控制器检测区域中的一个或多个麦克风和一个或多个扬声器1712，经由一个或多个麦克风测量由一个或多个扬声器在区域内生成的音频信号的初始频率响应，以及生成初始房间表现评级1714。该过程还可以包括将初始频率响应与目标频率响应进行比较1716，基于比较创建音频补偿值以应用于一个或多个扬声器1718，将音频补偿值应用于一个或多个扬声器1720，以及基于所应用的补偿值生成报告以识别优化的房间表现评级，而优化的房间表现评级产生比与初始房间表现评级相关联的音频表现值更优的一个或多个增强的音频表现值1722。FIG17 shows an example flow chart of a process for determining an initial audio profile of a room based on an ideal frequency response and attempting to modify the audio profile according to an example embodiment. Referring to FIG17 , the process may include detecting one or more microphones and one or more speakers in a zone via a controller 1712, measuring an initial frequency response of an audio signal generated by one or more speakers in the zone via the one or more microphones, and generating an initial room performance rating 1714. The process may also include comparing the initial frequency response to a target frequency response 1716, creating an audio compensation value based on the comparison to apply to one or more speakers 1718, applying the audio compensation value to the one or more speakers 1720, and generating a report based on the applied compensation value to identify an optimized room performance rating, and the optimized room performance rating produces one or more enhanced audio performance values 1722 that are better than the audio performance value associated with the initial room performance rating.

该过程还可以包括确定在音频呈现期间要占据区域的人员的预期密度，在将补偿值应用于一个或多个扬声器之前测量初始语音清晰度评分，以及基于所产生的初始语音清晰度评分确定所需的音频补偿值以实现由一个或多个扬声器产生的目标清晰度评分，该目标清晰度评分将适应人员的预期密度。确定要占据区域的人员的预期密度可以包括确定人员的可能位置，一个或多个扬声器包括位于区域的不同位置的两个或更多个扬声器，音频补偿值包括为两个或更多个扬声器中的每一者创建的相应两个或更多个扬声器优化值。该过程还可以包括将两个或更多个扬声器优化值应用于最靠近人员的可能位置的两个或更多个扬声器。该过程还可以包括当传感器检测到进入或离开区域的人数改变时，调整两个或更多个扬声器优化值。该过程还可以包括在将补偿值应用于一个或多个扬声器之后，经由一个或多个麦克风测量由一个或多个扬声器在区域内生成的经补偿的音频信号的经补偿的频率响应。该过程还可以包括将所测量的经补偿的频率响应与目标频率响应进行比较，以及确认所测量的补偿频率响应比初始频率响应更靠近目标频率响应值。The process may also include determining an expected density of people to occupy the area during the audio presentation, measuring an initial speech intelligibility score before applying the compensation value to one or more speakers, and determining the required audio compensation value based on the generated initial speech intelligibility score to achieve a target intelligibility score produced by the one or more speakers, the target intelligibility score will accommodate the expected density of people. Determining the expected density of people to occupy the area may include determining possible locations of people, the one or more speakers include two or more speakers located at different locations of the area, and the audio compensation value includes corresponding two or more speaker optimization values created for each of the two or more speakers. The process may also include applying the two or more speaker optimization values to the two or more speakers closest to the possible locations of the people. The process may also include adjusting the two or more speaker optimization values when the sensor detects a change in the number of people entering or leaving the area. The process may also include measuring a compensated frequency response of a compensated audio signal generated by the one or more speakers in the area via one or more microphones after applying the compensation value to the one or more speakers. The process may also include comparing the measured compensated frequency response with the target frequency response, and confirming that the measured compensated frequency response is closer to the target frequency response value than the initial frequency response.

在一个示例中，启动优化过程可以识别并且调整第一麦克风‘1’，第一麦克风‘1’具有34dB SPL A-加权的预启动噪声水平平均值以及23dB SPL A-加权的启动优化传输噪声水平平均值，其中所应用的噪声水平减小为‘低’。第二麦克风'2'可以具有34dB SPL A-加权的预启动噪声水平平均值以及24dB SPL A-加权的启动优化传输噪声水平平均值，其中所应用的噪声水平降低为‘低'。每个房间具有将影响扬声器表现的声学特征，并且调谐被需要以确保远端音频是清晰的并且所有用户可以在整个区域中最佳地听到音频。测量扬声器频率响应并且将(一个或多个)测量值与已知的表现值进行比较，以及对来自目标响应的变化启动自动补偿确保了该房间中的峰值表现。In one example, the startup optimization process may identify and adjust a first microphone '1' having a 34dB SPL A-weighted pre-startup noise level average and a 23dB SPL A-weighted startup optimized transmission noise level average, with the applied noise level reduced to 'low'. A second microphone '2' may have a 34dB SPL A-weighted pre-startup noise level average and a 24dB SPL A-weighted startup optimized transmission noise level average, with the applied noise level reduced to 'low'. Each room has acoustic characteristics that will affect the speaker performance, and tuning is required to ensure that the far-end audio is clear and that all users can hear the audio optimally throughout the area. Measuring the speaker frequency response and comparing the measured value(s) to known performance values, and enabling automatic compensation for variations from the target response ensures peak performance in the room.

结合本文公开的实施例描述的方法或算法的操作可以被直接体现在硬件中、由处理器执行的计算机程序中或两者的组合中。计算机程序可以包括在计算机可读介质上，例如存储介质。例如，计算机程序可以驻留在随机存取存储器(“RAM”)、闪存、只读存储器(“ROM”)、可擦除可编程只读存储器(“EPROM”)、电可擦除可编程只读存储器(“EEPROM”)、寄存器、硬盘、可移动盘、光盘只读存储器(“CD-ROM”)或本领域已知的任何其他形式的存储介质中。The operation of the method or algorithm described in conjunction with the embodiments disclosed herein may be directly embodied in hardware, in a computer program executed by a processor, or in a combination of the two. The computer program may be included on a computer-readable medium, such as a storage medium. For example, the computer program may reside in a random access memory ("RAM"), a flash memory, a read-only memory ("ROM"), an erasable programmable read-only memory ("EPROM"), an electrically erasable programmable read-only memory ("EEPROM"), a register, a hard disk, a removable disk, a compact disk read-only memory ("CD-ROM"), or any other form of storage medium known in the art.

图18并不旨在对本文描述的申请的实施例的使用范围或功能提出任何限制。无论如何，计算节点1800能够实现和/或执行上文阐述的任何功能。18 is not intended to impose any limitation on the scope of use or functionality of the embodiments of the application described herein. Regardless, computing node 1800 is capable of implementing and/or performing any of the functions set forth above.

在计算节点1800中，存在计算机系统/服务器1802，其可以与许多其他通用或专用计算系统环境或配置一起操作。可以适用于计算机系统/服务器1802的众所周知的计算系统、环境和/或配置的示例包括但不限于：个人计算机系统、服务器计算机系统、瘦客户端、富客户端、手持式或膝上型设备、多处理器系统、基于微处理器的系统、机顶盒、可编程消费电子产品、网络PC、小型计算机系统、大型计算机系统以及包括上述系统或设备中的任何一个的分布式云计算环境等。In computing node 1800, there is a computer system/server 1802, which can operate with many other general or special computing system environments or configurations. Examples of well-known computing systems, environments, and/or configurations that can be suitable for computer system/server 1802 include, but are not limited to: personal computer systems, server computer systems, thin clients, rich clients, handheld or laptop devices, multiprocessor systems, microprocessor-based systems, set-top boxes, programmable consumer electronics, network PCs, minicomputer systems, mainframe computer systems, and distributed cloud computing environments including any of the above systems or devices, etc.

可以在由计算机系统执行的计算机系统可执行指令的一般上下文(例如程序模块)中描述计算机系统/服务器1802。通常，程序模块可包括执行特定任务或实现特定抽象数据类型的例程、程序、对象、组件、逻辑、数据结构等。可以在分布式云计算环境中实践计算机系统/服务器1802，在分布式云计算环境中任务由通过通信网络链接的远程处理设备执行。在分布式云计算环境中，程序模块可以位于包括存储器存储设备的本地和远程计算机系统存储介质中。Computer system/server 1802 may be described in the general context of computer system executable instructions executed by a computer system, such as program modules. Generally, program modules may include routines, programs, objects, components, logic, data structures, etc. that perform specific tasks or implement specific abstract data types. Computer system/server 1802 may be practiced in a distributed cloud computing environment where tasks are performed by remote processing devices that are linked through a communications network. In a distributed cloud computing environment, program modules may be located in both local and remote computer system storage media including memory storage devices.

如图18所示，以通用计算设备的形式示出云计算节点1800中的计算机系统/服务器1802。计算机系统/服务器1802的组件可以包括但不限于：一个或多个处理器或处理单元1804、系统存储器1806、以及将包括系统存储器1806的各种系统组件耦合到处理器1804的总线。As shown in Figure 18, a computer system/server 1802 in a cloud computing node 1800 is shown in the form of a general-purpose computing device. The components of the computer system/server 1802 may include, but are not limited to, one or more processors or processing units 1804, a system memory 1806, and a bus that couples various system components including the system memory 1806 to the processor 1804.

总线表示任何若干种类型的总线结构中的一种或多种，包括存储器总线或存储器控制器、外围总线、加速图形端口以及使用各种总线架构中的任何一种的处理器或局部总线。作为示例而非限制，这种架构包括工业标准架构(ISA)总线、微声道架构(MCA)总线、增强型ISA(EISA)总线、视频电子标准协会(VESA)本地总线和外围组件互连(PCI)总线。Bus refers to one or more of any of several types of bus structures, including a memory bus or memory controller, a peripheral bus, an accelerated graphics port, and a processor or local bus using any of a variety of bus architectures. By way of example and not limitation, such architectures include an Industry Standard Architecture (ISA) bus, a Micro Channel Architecture (MCA) bus, an Enhanced ISA (EISA) bus, a Video Electronics Standards Association (VESA) local bus, and a Peripheral Component Interconnect (PCI) bus.

计算机系统/服务器1802通常包括各种计算机系统可读介质。这种介质可以是可由计算机系统/服务器1802访问的任何可用介质，并且它包括易失性和非易失性介质、可移动和不可移动介质。在一个实施例中，系统存储器1806实现其他附图的流程图。系统存储器1806可以包括易失性存储器形式的计算机系统可读介质，例如随机存取存储器(RAM)1810和/或缓冲存储器1812。计算机系统/服务器1802还可以包括其他可移动/不可移动、易失性/非易失性计算机系统存储介质。仅作为示例，可以提供存储系统1814用于从不可移动、非易失性磁介质(未示出并且通常被称为“硬盘驱动器”)读取并且向其写入。虽然没有示出，但是可以提供用于从可移动、非易失性磁盘(例如，“软盘”)读取并且向其写入的磁盘驱动器以及用于从可移动、非易失性光盘(例如，CD-ROM，DVD-ROM或其他光学介质)读取或向其写入的光盘驱动器。在这种实例中，每个可以通过一个或多个数据介质接口连接到总线。如下面将进一步描绘和描述的，存储器1806可以包括至少一个程序产品，该程序产品具有被配置为执行本申请的各种实施例的功能的一组(例如，至少一个)程序模块。Computer system/server 1802 typically includes various computer system readable media. Such media can be any available media that can be accessed by computer system/server 1802, and it includes volatile and non-volatile media, removable and non-removable media. In one embodiment, system memory 1806 implements the flow charts of other figures. System memory 1806 can include computer system readable media in the form of volatile memory, such as random access memory (RAM) 1810 and/or buffer memory 1812. Computer system/server 1802 can also include other removable/non-removable, volatile/non-volatile computer system storage media. By way of example only, a storage system 1814 can be provided for reading from and writing to a non-removable, non-volatile magnetic medium (not shown and commonly referred to as a "hard drive"). Although not shown, a disk drive for reading from and writing to a removable, non-volatile disk (e.g., a "floppy disk") and an optical drive for reading from or writing to a removable, non-volatile optical disk (e.g., CD-ROM, DVD-ROM or other optical media) can be provided. In this instance, each may be connected to the bus via one or more data media interfaces. As will be further depicted and described below, memory 1806 may include at least one program product having a set (eg, at least one) of program modules configured to perform the functions of various embodiments of the present application.

作为示例而非限制，可以在存储器1806中存储具有一组(至少一个)程序模块1818的程序/实用程序1816以及操作系统、一个或多个应用程序、其他程序模块和程序数据。操作系统、一个或多个应用程序、其他程序模块以及程序数据或其某种组合中的每一者可以包括联网环境的实施方式。程序模块1818通常执行本文描述的本申请的各种实施例的功能和/或方法。By way of example and not limitation, a program/utility 1816 having a set (at least one) of program modules 1818 and an operating system, one or more application programs, other program modules, and program data may be stored in memory 1806. Each of the operating system, one or more application programs, other program modules, and program data, or some combination thereof, may include an implementation of a networking environment. Program modules 1818 generally perform the functions and/or methods of various embodiments of the present application described herein.

如本领域技术人员将认识到的，本申请的各方面可以体现为系统、方法或计算机程序产品。因此，本申请的各方面可以是完全硬件实施例、完全软件实施例(包括固件、常驻软件、微代码等)或组合软件和硬件方面的实施例的形式，这些方面在本文中一般地都被称为“电路、“模块”或“系统”。此外，本申请的各方面可以采取被体现在一个或多个计算机可读介质中的计算机程序产品的形式，这些计算机可读介质上包含计算机可读程序代码。As will be appreciated by those skilled in the art, aspects of the present application may be embodied as a system, method, or computer program product. Thus, aspects of the present application may be in the form of a complete hardware embodiment, a complete software embodiment (including firmware, resident software, microcode, etc.), or a combination of software and hardware aspects, which are generally referred to herein as "circuits," "modules," or "systems." In addition, aspects of the present application may take the form of a computer program product embodied in one or more computer-readable media containing computer-readable program code.

计算机系统/服务器1802还可以与一个或多个外部设备1820(例如键盘、定点设备、显示器1822等)；使用户能够与计算机系统/服务器1802交互的一个或多个设备；和/或使计算机系统/服务器1802能够与一个或多个其他计算设备通信的任何设备(例如，网络卡、调制解调器等)通信。这种通信可以经由I/O接口1824发生。然而，计算机系统/服务器1802可以经由网络适配器1826与诸如局域网(LAN)、通用广域网(WAN)和/或公共网络(例如，互联网)之类的一个或多个网络通信。如图所示，网络适配器1826经由总线与计算机系统/服务器1802的其他组件通信。应当理解，虽然没有被示出，但是可以结合计算机系统/服务器1802使用其他硬件和/或软件组件。示例包括但不限于：微代码、设备驱动器、冗余处理单元、外部磁盘驱动器阵列、RAID系统、磁带驱动器和数据档案存储系统等。The computer system/server 1802 may also communicate with one or more external devices 1820 (e.g., keyboard, pointing device, display 1822, etc.); one or more devices that enable a user to interact with the computer system/server 1802; and/or any device that enables the computer system/server 1802 to communicate with one or more other computing devices (e.g., network card, modem, etc.). Such communication may occur via an I/O interface 1824. However, the computer system/server 1802 may communicate with one or more networks such as a local area network (LAN), a general wide area network (WAN), and/or a public network (e.g., the Internet) via a network adapter 1826. As shown, the network adapter 1826 communicates with other components of the computer system/server 1802 via a bus. It should be understood that, although not shown, other hardware and/or software components may be used in conjunction with the computer system/server 1802. Examples include, but are not limited to, microcode, device drivers, redundant processing units, external disk drive arrays, RAID systems, tape drives, and data archive storage systems, etc.

本领域技术人员将理解，“系统”可以被体现为个人计算机、服务器、控制台、个人数字助理(PDA)、蜂窝电话、平板计算设备、智能电话或任何其他合适的计算设备或设备的组合。将上述功能呈现为由“系统”执行并不旨在以任何方式限制本申请的范围，而是旨在提供许多实施例的一个示例。实际上，可以以与计算技术一致的本地化和分布式形式来实现本文公开的方法、系统和装置。Those skilled in the art will appreciate that a "system" may be embodied as a personal computer, server, console, personal digital assistant (PDA), cellular telephone, tablet computing device, smart phone, or any other suitable computing device or combination of devices. Presenting the above functions as being performed by a "system" is not intended to limit the scope of the present application in any way, but is intended to provide an example of many embodiments. In practice, the methods, systems, and apparatus disclosed herein may be implemented in localized and distributed forms consistent with computing technology.

应当注意，在本说明书中描述的一些系统特征被呈现为模块，以便更具体地强调它们的实施方式独立性。例如，模块可以被实现为硬件电路，该硬件电路包括定制超大规模集成(VLSI)电路或门阵列、现成的半导体，例如逻辑芯片、晶体管或其他分立组件。还可以在诸如现场可编程门阵列、可编程阵列逻辑、可编程逻辑设备、图形处理单元等之类的可编程硬件设备中实现模块。It should be noted that some system features described in this specification are presented as modules in order to more specifically emphasize their implementation independence. For example, a module can be implemented as a hardware circuit that includes custom very large scale integration (VLSI) circuits or gate arrays, off-the-shelf semiconductors such as logic chips, transistors, or other discrete components. Modules can also be implemented in programmable hardware devices such as field programmable gate arrays, programmable array logic, programmable logic devices, graphics processing units, etc.

还可以以用于由各种类型的处理器执行的软件来至少部分地实现模块。所识别的可执行代码的单元可以例如包括计算机指令的一个或多个物理或逻辑块，这些指令可以例如被组织为对象、程序或函数。然而，所识别的模块的可执行文本的物理位置不一定在一起，但是可以包括存储在不同位置中的不同指令，这些指令在逻辑地结合在一起时包括该模块并且实现该模块的所述目的。此外，模块可以被存储在计算机可读介质上，该计算机可读介质可以是例如硬盘驱动器、闪存设备、随机存取存储器(RAM)、磁带或用于存储数据的任何其他这种介质。Modules can also be implemented at least in part with software for being executed by various types of processors. The unit of the executable code identified can, for example, include one or more physical or logical blocks of computer instructions, which can, for example, be organized as objects, programs or functions. However, the physical location of the executable text of the identified module is not necessarily together, but can include different instructions stored in different locations, which include the module and realize the described purpose of the module when logically combined together. In addition, modules can be stored on a computer-readable medium, which can be, for example, a hard disk drive, a flash memory device, a random access memory (RAM), a magnetic tape or any other such medium for storing data.

实际上，可执行代码的模块可以为单个指令或许多指令，并且可以甚至被分布在若干不同的代码段上、在不同的程序之间、以及在若干存储设备之上。类似地，可操作的数据在本文中可以被识别或示出在模块内，并且可以以任何适当形式被体现并且被组织在任何适当类型的数据结构内。可操作的数据可以被收集为单个数据集，或可以被分布在不同的位置上(包括不同的存储设备上)，并且可以至少部分地仅仅作为电信号存在于系统或网络上。In fact, the module of executable code can be a single instruction or many instructions, and can even be distributed on several different code segments, between different programs, and on several storage devices. Similarly, operable data can be identified or shown in the module herein, and can be embodied in any appropriate form and organized in the data structure of any appropriate type. Operable data can be collected as a single data set, or can be distributed in different locations (including different storage devices), and can exist on a system or network at least in part only as an electrical signal.

将容易理解的是，可以以各种不同的配置布置和设计在本文附图中总体描述和说明的本申请的组件。因此，对实施例的详细描述并不旨在限制所要求保护的申请的范围，而仅仅代表本申请的所选实施例。It will be readily understood that the components of the present application generally described and illustrated in the drawings herein may be arranged and designed in a variety of different configurations. Therefore, the detailed description of the embodiments is not intended to limit the scope of the claimed application, but merely represents selected embodiments of the present application.

本领域的普通技术人员将容易理解，可以以不同顺序的步骤和/或用与所公开的配置不同的配置中的硬件元件来实践上述内容。因此，虽然已经基于这些优选实施例描述了本申请，但是对于本领域的技术人员显而易见的是，某些修改、变化和替代构造将是显而易见的。Those skilled in the art will readily appreciate that the above may be practiced in different orders of steps and/or with hardware elements in configurations different from those disclosed. Therefore, although the present application has been described based on these preferred embodiments, it will be apparent to those skilled in the art that certain modifications, variations, and alternative configurations will be apparent.

虽然已经描述了本申请的优选实施例，但是应当理解，所描述的实施例仅仅是说明性的，并且当考虑到其全部范围的等同物和修改(例如，协议、硬件设备、软件平台等)时，本申请的范围仅由所附权利要求来限定。Although preferred embodiments of the present application have been described, it should be understood that the described embodiments are merely illustrative and that the scope of the present application is limited solely by the appended claims when considering its full range of equivalents and modifications (e.g., protocols, hardware devices, software platforms, etc.).

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