"...Early psychoacoustic research suggested that the human auditory system is insensitive to differences in the relative phases of spectral components of a multicomponent sound. "......早期的心理声学研究表明，人类听觉系统对多成分声音的光谱成分的相对相位差异并不敏感。 However, research from the last two decennia provides evidence that listeners can detect phase differences between the stimulus components that interact within a single auditory filter. 然而，过去二十年的研究证明，听者可以检测到在单个听觉滤波器中相互作用的刺激成分之间的相位差。 The most impressive demonstration of phase sensitivity is given by the masker-phase effect, i.e. the more than variation in masking effect caused by a harmonic complex when varying the phase relations between its components. This masking paradigm is widely used to obtain a psychoacoustical measure of the phase response of the cochlea...." 掩蔽器相位效应是相位敏感性最令人印象深刻的证明，即当谐波复合物的各组成部分之间的相位关系发生变化时，其掩蔽效应会发生 以上的变化。这种掩蔽范例被广泛用于获得耳蜗相位响应的心理声学测量...."
I must admit, I did not know about the above research and it's results. I have been researching the internet for about a year before I came across the above, simple information. There is a lot more too. 我必须承认，我并不知道上述研究及其结果。我在互联网上搜索了大约一年，才发现了上述这些简单的信息。还有很多其他的信息。 Now, I realised, there is a volume of research results that clearly indicates, that rather than asking "is phase distortion audible?" we should now be asking question "how does the phase distortion manifest itself?". 现在，我意识到，大量的研究成果清楚地表明，与其问 "相位失真听得见吗？"，不如问 "相位失真是如何表现出来的？"。
Naively, and without any prior experience in how should I actually do it, I conducted my own listening tests by comparing the sound from traditional, minimumphase loudspeakers to the sound of linear-phase loudspeakers. 在没有任何经验的情况下，我天真地将传统最小相位扬声器的声音与线性相位扬声器的声音进行了比较，并进行了自己的听音测试。 I am talking here about acoustically linear-phase loudspeaker. During my short, initial listening tests on linear-phase loudspeakers, I was surprised by how indifferent the linear-phase mode was to my ear. 我这里说的是线性相位扬声器。在我对线性相位扬声器进行的简短、初步试听测试中，线性相位模式对我的耳朵来说是如此漠不关心，令我大吃一惊。
Was I doing the right thing then?. This result definitely required further investigation on much more diverse listening material. 那我做得对吗？这一结果无疑需要在更多样化的听力材料上做进一步的研究。
Listening Habits 聆听习惯
Traditionally, when I listened to the quality of the sound reproduced by my audio playback equipment, I focus on tonal balance (frequency response), dynamics of the sound (SNR), residual noise floor ( inaudible ), distortion ( inaudible ). 传统上，当我聆听音频回放设备重现的声音质量时，我会关注音调平衡（频率响应）、声音动态（信噪比）、本底残余噪声（听不见）、失真（听不见）。
Interestingly, all of the above characteristics can be assessed and visualized in frequency domain. 有趣的是，上述所有特征都可以在频域中进行评估和可视化。 It was simply the easiest way to listen to the sound and evaluate what I was hearing, but I now realize, that I was only considering the steady-state analysis in the frequency domain - see pictures below. 这只是聆听声音和评估我所听到的声音的最简单方法，但我现在意识到，我只考虑了频域的稳态分析--见下图。
Frequency response, distortion, dynamics and noise floor - all in frequency domain. 频率响应、失真、动态和本底噪声--全部都在频域内。
I was doing the same type of analysis over, and over again for years, and grew accustomed to this ritual. It was easy to compare with measured results, so it felt comfortable, that I can correlate my measurements with what I can easily hear (or can not hear). 多年来，我一遍又一遍地做着同一类分析，渐渐习惯了这种仪式。这很容易与测量结果进行比较，因此我感觉很舒服，因为我可以将测量结果与我很容易听到（或听不到）的声音联系起来。
Recently, things have changed for me. I came across a simple paper, http://www.audiophilerecordingstrust.org.uk/articles/speaker_science.pdf which inspired me to take a more comprehensive look at my listening tests. 最近，我的情况发生了变化。我看到了一篇简单的论文http://www.audiophilerecordingstrust.org.uk/articles/speaker_science.pdf，它启发我更全面地审视我的听力测试。 Having read the paper, I re-examined information from other internet re-sources, and as a result I came to the conclusion, that my listening tests were only a starting point of what I should have listened to when examining linear-phase loudspeakers. 读完这篇论文后，我重新研究了其他互联网资料来源的信息，得出的结论是，我的听力测试只是我在研究线性相位扬声器时应该聆听的一个起点。
To put it simply - I needed to significantly extended the evaluation of timedomain characteristics of the loudspeaker in my listening habits. 简而言之，在我的听音习惯中，我需要大大扩展对扬声器时域特性的评估。
In the brief conclusions of my short, initial listening tests presented in http://www.bodziosoftware.com.au/Home Theatre_Conclusions.pdf I have pointed out one perceptible difference - I felt closer to the stage/musicians. This was more of an accidental and unexpected impression, to which I did not pay much attention. 在 http://www.bodziosoftware.com.au/Home Theatre_Conclusions.pdf 中，我对自己简短的初步试听进行了简要总结，指出了一个明显的不同之处--我感觉更接近舞台/音乐家。这更多的是一种偶然和意外的印象，我并没有太在意。 But this indeed relates to time-domain characteristics of a loudspeaker, rather than frequency domain. 但这确实与扬声器的时域特性有关，而不是频域特性。
Yes, it appears, that I have been covering only half of what I should have been paying attention to. And the paper mentioned above made it startlingly clear to me. 是的，看来我只关注了我应该关注的一半。上面提到的那篇论文让我豁然开朗。
New Listening Habits 新的聆听习惯
The remaining part of this paper is my crude attempt to summarise audible attributes of linear-phase loudspeakers. This is what you need to listen for when evaluating linear-phase loudspeakers. I do not pretend, that the list is complete, but it's a start. 本文的其余部分是我对线性相位扬声器可听特性的粗略总结。这是您在评估线性相位扬声器时需要聆听的内容。我并不妄言这份清单是完整的，但这只是一个开始。 It clearly points to the time-domain characteristics of the loudspeaker, and this is something, which may of us (till recently, including myself) are not accustomed to. I simply did not know what to listen for. 它清楚地指出了扬声器的时域特性，而这是我们（直到最近，包括我自己）可能还不习惯的。我根本不知道该听什么。
Below, I present the "nominated attribute(s)", showing the source, followed by a short description from the source. 下面，我将介绍 "提名属性"，显示来源，然后是来源的简短说明。
1. Tighter bass 1.更紧凑的低音
2. Wider and deeper sound stage (quite dramatic) 2.更宽更深的声场（颇具戏剧性）
"...A highlight this year was a demo of the capabilities of DEQX. This came about from discussions of my active crossover listening comparisons, in which a small group could not hear any improvement with DEQX. "......今年的一大亮点是演示 DEQX 的功能。这源于对我的有源分频器试听对比的讨论，其中有一小部分人听不出 DEQX 有任何改进。 Terry argued that we had dumbed down the DEQX and prevented it from showing what it can do. This is certainly true, we wanted to test sound quality only and in that regard found no reason to spend the extra compared to cheaper options. Terry 认为我们降低了 DEQX 的音质，使其无法展现出应有的功能。这当然是对的，我们只想测试音质，在这方面，我们认为没有理由比便宜的选择多花钱。 However, Terry set up a demo in which two profiles were created on DEQX. One was limited to the processing power of MiniDSP and DCX. The other allowed DEQX to strut its stuff. In particular, it was allowed to correct for phase and group delay. 不过，Terry 设置了一个演示，在 DEQX 上创建了两个配置文件。其中一个仅限于 MiniDSP 和 DCX 的处理能力。另一个则允许 DEQX 大显身手。特别是，它可以校正相位和群延迟。 We then blind tested this with instant switching, not knowing what was being heard. I was the first to sit in the chair and do the demo and quite soon I didn't need to be told which was which, because the difference was obvious. 然后，我们进行了盲测，即时切换，不知道听到的是什么。我是第一个坐在椅子上进行演示的人，很快我就不需要别人告诉我哪个是哪个了，因为区别是显而易见的。
Changes noticed with DEQX: 注意到 DEQX 的变化：
much tighter bass 更紧凑的低音
wider and deeper sound stage (quite dramatic) 更宽阔、更深沉的声场（相当震撼）
Both had a basic level of time alignment with digital delays. Both were matched in level and in response closely. These differences were related to the group delay correction. Without it, the sound was flat and almost lifeless in comparison. 两者都具有基本的数字延迟时间一致性。两者在电平和响应方面都非常匹配。这些差异与群延迟校正有关。如果没有群延迟校正，声音就会显得平淡无奇，几乎没有生命力。
I then watched as others sat through the demo, each person noticing the same differences, differing only in the amount of time taken before declaring what they heard....." 我随后观看了其他人的演示，每个人都注意到了相同的差异，不同的只是在宣布他们听到了什么之前所花的时间....."。
Personally, I can testify to the tighter bass audible during linear-phase mode. I operate large, vented enclosure subwoofers, tuned to 20 Hz . Playing impulsive sounds, in minimum-phase mode, the subs overshot and then add and prolong the ringing - past steep, impulse-like signals. 就我个人而言，我可以证明在线性相位模式下可以听到更紧凑的低音。我使用的是大型的 通风外壳低音炮，调谐频率为 20 Hz。在最小相位模式下播放脉冲信号时，低音炮会超音，然后增加并延长振铃--经过陡峭的脉冲信号。 This unwanted flabbiness is unfortunately audible in minimum-phase mode on low-frequency impulsive signals. 遗憾的是，在最小相位模式下，低频脉冲信号会出现这种不必要的松弛现象。
(This is a must-read article in it's entirety) (此文全文必读）
".....Another area in which loudspeakers are disreputable is in the neglect of the time domain. The traditional view is that all that matters is to be able to reproduce continuous sine waves over the range of human hearing. "..... "扬声器令人不齿的另一个方面是对时域的忽视。传统观点认为，重要的是能够在人类听觉范围内重现连续的正弦波。
A very small amount of research and thought will reveal that this is a misguided view. Frequency response is important, but not so important that the attainment of an ideal response should be to the detriment of realism. 只要稍加研究和思考，就会发现这种观点是错误的。频率响应固然重要，但还不至于为了获得理想的响应而损害现实性。 One tires of hearing that "phase doesn't matter" in audio or "the ear is phase deaf". These are outmoded views which were reached long ago in flawed experiments and which are at variance with the results of recent psychoacoustic research. 音频中 "相位无关紧要 "或 "耳朵是相位耳聋 "的说法已经听腻了。这些都是过时的观点，是很久以前在有缺陷的实验中得出的，与最近的心理声学研究结果不符。
The ear works in two distinct ways, which it moves between in order to obtain the best outcome from the fundamental limits due to the Heisenberg inequality. The Heisenberg inequality states that as frequency resolution goes up, time resolution goes down and vice versa. 由于海森堡不等式的存在，为了从基本极限中获得最佳效果，耳朵会以两种不同的方式工作。海森堡不等式指出，频率分辨率越高，时间分辨率越低，反之亦然。 Real sounds are not continuous, but contain starting transients. During such transients, the ear works in the time domain. 真实的声音不是连续的，而是包含起始瞬态。在这种瞬态期间，耳朵在时域中工作。 Before the listener is conscious of a sound, the time domain analysis has compared the time of arrival of the transient at the two ears and established the direction. 在听者意识到声音之前，时域分析已经比较了瞬态到达两耳的时间，并确定了方向。 Following the production of a transient pressure step by a real sound source, the sound pressure must equalise back to ambient. 在实际声源产生瞬态压力阶跃后，声压必须恢复到环境温度。
The rate at which this happens is a function of the physical size of the source. The ear, again acting in the time domain, can measure the relaxation time and assess the size of the source. 发生这种情况的速度与声源的物理大小有关。同样是在时域中，耳朵可以测量弛豫时间并评估声源的大小。 Thus before any sound is perceived, the mental model has been told of the location and size of a sound source. 因此，在感知到任何声音之前，心理模型就已经知道了声源的位置和大小。
In fact this was the first use of hearing, as a means of perceiving a threat in order to survive. Frequency analysis in hearing, consistent with the evolution of speech and music came much later. 事实上，这是听觉的最初用途，是为了生存而感知威胁的一种手段。听觉中的频率分析，与语言和音乐的进化相一致，则要晚得多。 After the analysis of the initial transient, the ear switches over to working in the frequency domain in order to analyses timbre. In this mode, the mode that will be used on steady state signals, phase is not very important. 在对初始瞬态信号进行分析之后，耳朵会切换到频域工作模式，以分析音色。在这种模式下，也就是用于稳态信号的模式下，相位并不十分重要。 However, the recognition of the initial transient and the relaxation time are critical for realism. Anything in a sound reproduction system which corrupts the initial transient is detrimental. 然而，对初始瞬态和松弛时间的识别对于逼真度至关重要。声音再现系统中任何破坏初始瞬态的因素都是有害的。
Whilst audio electronics can accurately handle transients, the traditional loudspeaker destroys both the transient and the relaxation time measurement. 虽然音频电子设备可以准确处理瞬态，但传统的扬声器会破坏瞬态和松弛时间的测量。 Lack of attention to the time domain in crossover networks leads to loudspeakers which reproduce a single input step as a series of steps, one for each drive unit at different times..." 对分频网络中的时域缺乏关注，导致扬声器将单个输入阶跃重现为一系列阶跃，每个驱动单元在不同时间都有一个阶跃......"
Boston Audio Society has an interesting view on time-corrected loudspeakers. 波士顿音频协会对时间校正扬声器的看法很有意思。
"....If the stereo loudspeakers differ in their time-shift behaviour by more than about thirty millionths of a second (or a finer tolerance, perhaps, for critical listeners), the stereo image will be perceptibly smeared. "....，如果立体声扬声器在时移行为上的差异超过大约三千万分之一秒（对于挑剔的听众来说，容差可能会更小），立体声图像就会明显模糊。 The two speakers must "speak" together at all frequencies if the subtlest details in the stereo field are to be preserved. 如果要保留立体声声场中最微妙的细节，两个扬声器必须在所有频率上一起 "说话"。
This, quite simply, may be the principal advantage to be gained from "linear-phase" or "time-corrected" loudspeakers. 简单地说，这可能是 "线性相位 "或 "时间校正 "扬声器的主要优势。 The manufacturers who are striving to reduce the time dispersion of loudspeakers to zero may also be ensuring that there will be no significant differences in signal propagation timing between the two speakers in a stereo pair. 努力将扬声器的时间频散减小到零的制造商，可能也是为了确保在一对立体声扬声器中，两个扬声器之间的信号传播时间不会有明显差异。 The delicate timing information in a stereo recording is thus accurately retained and is transmitted to the listener unaltered..." 因此，立体声录音中微妙的时间信息被准确地保留下来，并原封不动地传送给听众......"
They also point to some of the advantages of such loudspeakers: 他们还指出了此类扬声器的一些优点：
1. Depth. 1.深度
This may surprise some listeners when they first hear it, since many speakers (and records) elicit only a general left-to-right spread. 这可能会让一些听众在第一次听到时感到惊讶，因为许多扬声器（和唱片）只能引起一般的从左到右的传播。 But "stereo", as originally conceived, implied a three-dimensional sound in which voices or instruments could be localized at different apparent distances from the listener as well as at various lateral positions. 但是，"立体声 "最初的概念意味着一种三维声音，其中的声音或乐器可以定位在距离听众不同的明显距离以及不同的横向位置上。 Listeners to time-aligned speakers consistently report hearing a stereo image with unusual depth. 使用时间对齐扬声器的听众一致表示，他们听到的立体声图像具有不同寻常的深度。
2. Resolution. 2.决议。
The stereo image is reproduced precisely, each voice or instrument having its proper place and width. In complex sound sources such as symphony orchestra, individual instruments can be resolved with unexpected clarity. 立体声图像得到精确再现，每个声音或乐器都有其适当的位置和宽度。在交响乐团等复杂音源中，单个乐器的清晰度出乎意料。 In the old cliche, "I hear details I never knew were in the recording. 用一句老生常谈的话来说，"我听到了录音中从未有过的细节。 " Some listeners have incorrectly attributed the improved resolution of detail to more accurate transient response, but the better definition of details is simply the result of the reduction of blending in the stereo image. "一些听众错误地将细节分辨率的提高归因于更准确的瞬态响应，但细节的清晰度提高只是立体声图像中混合减少的结果。
3.Separation of ambience. 3.氛围分离
With loudspeakers whose stereo image is slightly blended because of time-smear, any hall ambience or reverberation in the recording tends to become slightly mixed with the instrumental sounds, causing coloration of those sounds. 如果扬声器的立体声图像因时差而略有混杂，则录音中的任何大厅氛围或混响往往会与乐器声略有混杂，从而使这些声音产生色彩。 Consequently, with such speakers closely-microphoned recordings tend to sound better because of their distinctly defined sound. But with time-corrected loudspeakers, the ambience is resolved as a separate sound, and larger amounts of hall ambience in recordings can be enjoyed. 因此，在使用此类扬声器时，近距离麦克风录音的声音往往会更好听，因为它们的音色清晰明了。但使用时间校正扬声器时，环境音可作为一种独立的声音得到解决，因此可以欣赏到录音中更多的大厅环境音。
7. Inter-channel accuracy of sound reproduction. 7.声音再现的信道间准确性。
Source: 资料来源
".......5. Audibility of Phase Distortion ".......5.相位失真的可听性
One of the confusing issues regarding the audibility of phase is that the discussion is generally considered to be a single topic when in reality should be discussed as two distinct situations. The audibility of phase distortion must be evaluated as follows: 有关相位可听度的一个令人困惑的问题是，人们通常认为讨论的是一个单一的主题，而实际上应该讨论两种不同的情况。相位失真的可听性必须按以下方式进行评估：
Inter-channel phase distortion. Characterized as differences in phase response between two or more channels. 通道间相位失真。以两个或多个通道之间的相位响应差异为特征。
Intra-channel phase distortion. Characterized by non-linear phase response within a channel with the stipulation that the phase response is matched between all channels within the system (i.e. inter-channel phase distortion is equal to 0 msec ) 信道内相位失真。特点是信道内的非线性相位响应，但系统内所有信道之间的相位响应必须匹配（即信道间相位失真等于 0 毫秒）。
6. Inter-Channel Phase Distortion 6.通道间相位失真
We use the amplitude and phase relationship between the sounds received by our ears to localize the source of the sound. 我们利用耳朵接收到的声音之间的振幅和相位关系来定位声源。 Modern audio systems use this attribute to create what is known as imaging, or the perception that an instrument or vocal is coming from a location that is different than the actual speaker location. 现代音频系统利用这一特性创造出所谓的 "声像"，即让人感觉到乐器或人声来自与实际扬声器位置不同的地方。 The audible effects of inter-channel phase distortion can be easily demonstrated by simply reversing the speaker connections on one channel of an otherwise properly configured stereo system. The loss of imaging is immediately noticeable even to those without a trained ear. 只要将一个配置正确的立体声系统中一个声道上的扬声器连接反接，就能很容易地证明声道间相位失真的声音效果。即使没有受过专业训练的耳朵，也能立即察觉到成像的损失。 Granted this test is rather dramatic and 180 degrees of inter-channel phase distortion is not indicative of standard operation but it does demonstrate the potential effects. 诚然，这个测试相当引人注目，180 度的通道间相位失真并不能说明标准操作，但它确实显示了潜在的影响。 As a result of this test, you would be hard pressed to find someone that would argue that 180 degrees of inter-channel phase distortion is acceptable, but where between the two extremes is the threshold of audibility? 测试结果表明，很难有人会认为 180 度的信道间相位失真是可以接受的，但两个极端之间的可听度阈值在哪里呢？ Tom Holman reports [10] that in his laboratory environment at the University of Southern California that is dominated by direct sound, a channel-to-channel time offset equal to one sample period at 48 kHz is audible. This equates to of inter-channel phase distortion across the entire audio band. Holman [10] also mentions, "one just noticeable difference in image shift between left and right ear inputs is ". 汤姆-霍尔曼（Tom Holman）报告说[10]，在他位于南加州大学的实验室环境中，以直达声为主，在 48 kHz 的频率下，可以听到相当于一个采样周期的通道间时间偏移。这相当于 整个音频频段的通道间相位失真。Holman [10] 还提到，"左右耳输入图像偏移的一个明显差别是 "。
7. Intra-Channel Phase Distortion 7.通道内相位失真
Recall that we use the differences in signal amplitude and phase to localize or determine the source of sound and relatively small amounts of inter-channel phase distortion can be audible. 回想一下，我们利用信号振幅和相位的差异来定位或确定声源，而相对较小的信道间相位失真也是可以听到的。 But how does our hearing react when each channel in a multi-channel system is subjected to non-linear phase response but the phase response is matched between all channels? 但是，当多通道系统中的每个通道都受到非线性相位响应的影响，但所有通道之间的相位响应都相匹配时，我们的听力会有什么反应呢？ Douglas Preis [11] did an extensive survey of existing literature and Tom Holman's [10] experiences and research through his work at USC gives us an interesting insight into this phenomenon. Douglas Preis [11]对现有文献进行了广泛调查，Tom Holman [10]在南加州大学工作的经历和研究也让我们对这一现象有了有趣的认识。 Both report that the threshold of audibility is frequency dependent, which correlates with all other audibility thresholds. 这两份报告都指出，可听阈值与频率有关，与所有其他可听阈值相关。 In laboratory environments when using test tones and headphones, research has shown that the human ear is sensitive to intra-channel phase differences of 0.25 msec [8] or [9] in the mid-range with the threshold increasing at higher and lower frequencies. Preis states "the tolerances shown.... are not directly applicable to speech or music signals irradiated by loudspeakers in a 在实验室环境中使用测试音和耳机时，研究表明人耳对 0.25 毫秒 [8] 或 [9] 的中频内相位差很敏感，阈值随频率的升高和降低而增大。Preis 指出：".... 所显示的公差并不直接适用于扬声器辐照的语音或音乐信号。
reverberant environment. Most likely, the perceptual thresholds for these conditions would be at more than twice those shown". 混响环境。这些条件下的感知阈值很可能是所示阈值的两倍以上"。 Essentially, the data suggests that for high quality music or speech reproduction in a reverberant environment intra-channel phase distortion of 1 msec is inaudible to a trained listener. 从本质上讲，这些数据表明，在混响环境中重现高质量音乐或语音时，1 毫秒的通道内相位失真对于训练有素的听者来说是听不到的。 Notice that this threshold is a relatively conservative statement and is still two orders of magnitude greater than that for inter-channel phase distortion!....." 请注意，这个阈值是一个相对保守的说法，仍然比信道间相位失真阈值大两个数量级！....."
8. Precedence effect or "law of the first wavefront" 8.先验效应或 "第一波面定律"
"....The precedence effect or law of the first wavefront is a binaural psychoacoustic effect. "....，先波面效应或先波面定律是一种双耳心理声学效应。 When a sound is followed by another sound separated by a sufficiently short time delay (below the listener's echo threshold), listeners perceive a single fused auditory image; its perceived spatial location is dominated by the location of the first-arriving sound (the first wave front). 当一个声音与另一个声音相隔的时间延迟足够短（低于听者的回声阈值）时，听者就会感知到一个融合的听觉图像；其感知的空间位置主要由最先到达的声音（第一波前沿）的位置决定。 The lagging sound also affects the perceived location. However, its effect is suppressed by the first-arriving sound..... 滞后声音也会影响感知位置。但是，它的影响会被先到的声音所抑制.....
The precedence effect appears, if the subsequent wave fronts arrive between 2 ms and about 50 ms later than the first wave front. 如果随后的波前到达时间比第一个波前晚 2 毫秒到 50 毫秒左右，就会出现优先效应。
The precedence effect is important for the hearing in enclosed rooms. With the help of this effect it remains possible to determine the direction of a sound source (e.g. the direction of a speaker) even in the presence of wall reflections...." 先声效果对封闭房间内的听觉非常重要。借助这种效应，即使在墙壁反射的情况下，也能确定声源的方向（如扬声器的方向）...."
"....Since Helmholtz, there has been a figurative tug-of-war between proponents of his "spectral theory" of musical sound and researchers who recognized the importance of sound's temporal properties. "....，自赫尔姆霍兹以来，他的音乐声音 "频谱理论 "的支持者与认识到声音时间特性重要性的研究人员之间一直存在着一场形象的拉锯战。 Analysis-by-synthesis research, by trying to discover methods for synthesizing realistic sounds, has revealed several critical limitations of purely spectral theories. 通过合成分析的研究，试图发现合成逼真声音的方法，揭示了纯光谱理论的几个关键局限。 Clark demonstrated that recordings played in reverse-which have the same magnitude spectra as their normal counterparts-make sound-source identification very difficult. 克拉克证明，反向播放的录音具有与正常录音相同的幅度光谱，这使得声源识别非常困难。 Synthesis based on Fourier spectra, with no account of phase, does not produce realistic sounds, in part because the onset properties of the sound are not captured (Clark et al., 1963). 基于傅立叶频谱的合成不考虑相位，无法产生逼真的声音，部分原因是没有捕捉到声音的起始特性（Clark 等人，1963 年）。 Although most musical instruments produce spectra that are nearly harmonic - that is, the frequencies of their components (measured in small time windows) are accurately modeled by integer multiples of a fundamental-deviations from strict harmonicity are critical to the sounds produced by some instruments. 虽然大多数乐器产生的频谱接近谐波，也就是说，其各组成部分的频率（在小时间窗内测量）都是以基音的整数倍来精确建模的，但与严格谐波性的偏差对某些乐器产生的声音至关重要。 For example, components of piano tones below middle-C (261 例如，中 C 以下钢琴音调的成分（261
Hz ) must be inharmonic to sound piano-like (Fletcher et al., 1962). Hz ）必须是非谐波，才能发出类似钢琴的声音（Fletcher 等人，1962 年）。 In fact, all freely vibrating strings (e.g., plucked, struck, or released from bowing) and bells produce inharmonic spectra, and inharmonicity is important to the attack of many instrument sounds (Freedman, 1967; Grey & Moorer, 1977). 事实上，所有自由振动的弦乐器（如拨弦、击弦或弓弦释放）和铃铛都会产生非谐波频谱，而非谐波性对许多乐器的音色攻击都很重要（Freedman, 1967; Grey & Moorer, 1977）。 Without erratic frequency behavior during a note's attack, synthesized pianos sound as if they have hammers made of putty (Moorer & Grey, 1977). 如果在音符的打击过程中没有不稳定的频率行为，合成钢琴听起来就像是用油灰做的锤（Moorer & Grey，1977 年）。
So Helmholtz's theory is correct as far as it goes: the relative phases of the components of a purely periodic sound matter little to perception. However, as soon as musical tone varies over time - for example, by turning on or off temporal properties become relevant. 因此，亥姆霍兹的理论就其本身而言是正确的：纯周期性声音各组成部分的相对相位与感知关系不大。然而，一旦乐音随时间发生变化，例如通过开启或关闭，时间特性就会变得相关。 In the real world, there are no purely periodic sounds, and an instrument's magnitude spectrum is but one of its facets .." 在现实世界中，并不存在纯粹的周期性声音，乐器的音阶谱只是其中的一个方面......"
10. Pitch, Timbre, and Source Separation 10.音高、音色和音源分离
"Near", "Far", and Harmonic Coherence 近"、"远 "和谐波相干性
Humans can immediately hear if a sound is "near" or "far" with a single ear. 人类只需一只耳朵就能立即听出声音是 "近 "还是 "远"。
But how do we perceive it, and how can it be measured? 但我们如何看待它，又如何衡量它呢？
The author believes that engagement, near/far, pitch perception, timbre perception, direction detection, and stream formation all derive from the same property of sound 作者认为，接触、近/远、音高感知、音色感知、方向探测和音流形成都源自声音的同一特性
the phase coherence of harmonics in the vocal formant range, to 4000 Hz . 声带（ 至 4000 Hz）谐波的相位一致性。
Example: The syllables one to ten with four different degrees of phase coherence. The sound power and spectrum of each group is identical 举例说明：一至十音节有四种不同程度的相位一致性。每组的声功率和频谱相同
11. Confirmation of two-stage processing by the ear, as discussed in (3). 11.如（3）所述，确认耳朵的两阶段处理。
4.3.1 The "law of the first localisation stimulus" 4.3.1 "第一次定位刺激定律"
"....For a conventional stereo-up, a phantom source shifts from to if the time difference between two broadband loudspeaker signals is increased from zero to about . The association model could explain this phenomenon (time- as well as level-based stereophony) by means of psychoacoustic principles of the gestalt association stage. "....，在传统的立体声中，如果两个宽带扬声器信号之间的时间差从零增加到 左右，幻象声源就会从 移动到 。联想模型可以通过格式塔联想阶段的心理声学原理来解释这种现象（基于时间和电平的立体声）。 The localisation stimulus arriving at the gestalt association stage first has a greater weight compared to the second stimulus (the equivalent for level based stereophony would be the localisation stimulus with the higher level). 与第二个刺激相比，首先到达格式联想阶段的定位刺激的权重更大（对于基于水平的立体声来说，等效于水平较高的定位刺激）。 Despite their identity and relative time delay, the localisation stimuli can be discriminated, since each of them is present in the binaural correlation pattern in a complete and discriminable form (see Section 4.1). 尽管定位刺激具有同一性和相对时间延迟性，但由于它们在双耳相关模式中都以完整和可分辨的形式存在（见第 4.1 节），因此还是可以分辨出来的。
Yet, a further increase in the inter-channel time difference leads to an exceedance of the maximal time delay max. For stationary broadband signals (continuous noise), this causes a disruption of the localisation stimulus selection, which manifests itself in the form of a reduced suppression of the comb filter effect, for example. 然而，信道间时差的进一步增大会导致超过最大时延 max。对于静止的宽带信号（连续噪声），这会导致定位刺激选择中断，表现为梳状滤波器效应的抑制减弱等。 In this particular sound field constellation, the law of the first wavefront cannot be observed in accordance with the association model. Analysable wavefronts that would allow for a localisation stimulus selection of the impinging sound components do not exist. 在这种特殊的声场星座中，无法根据关联模型观察到第一波面的规律。不存在可分析的波阵面，因此无法对撞击声成分进行定位刺激选择。
In contrast, for non-stationary impulsive signals (clicks, speech, impulsive tones) an increase in the inter-channel time difference has a different effect. 相反，对于非稳态脉冲信号（咔嗒声、语音、脉冲音），增加信道间时差会产生不同的效果。 In the association model, evaluation of the amplitude envelope ensures that the primary and the delayed sound (reflection) can be discriminated as localisation stimuli. 在联想模型中，对振幅包络线的评估可确保原声和延迟声（反射声）作为定位刺激得以区分。 According to a hypothetical function of the gestalt association stage, the primary localisation stimulus determines the auditory event. It does this even more so the larger the time difference between the arriving localisation stimuli gets. Only when a time difference of about is exceeded will the subsequent localisation stimulus gain in perceptual weight. Beyond the echo threshold (for a definition see BLAUERT 1974), it will be perceived as a separate auditory event. 根据格式塔联想阶段的假设功能，主要定位刺激决定了听觉事件。到达的定位刺激之间的时间差越大，它的作用就越大。只有当时间差超过 左右时，随后的定位刺激才会获得知觉权重。超过回声阈值（定义见 BLAUERT 1974）时，它将被视为一个独立的听觉事件。
It appears that the "law of the first wavefront" can be interpreted as the "law of the first localisation stimulus"...." 看来，"第一波面定律 "可以解释为 "第一定位刺激定律"...."
".....6. Summary ".....6.摘要
According to the association model presented in the preceding chapters, the functioning of the auditory system with respect to spatial hearing is due to two different processing mechanisms. Each of these two processing mechanisms manifests itself in the form of an associatively guided pattern selection. 根据前几章介绍的联想模型，听觉系统在空间听觉方面的功能是由两种不同的处理机制造成的。这两种处理机制分别以联想引导模式选择的形式表现出来。
A current stimulus stemming from a sufficiently broadband sound source gives rise to a location association in the first and to a gestalt association in the second, higher-level processing stage because of auditory experience. 源于足够宽带声源的电流刺激会在第一加工阶段产生位置联想，并在第二加工阶段，即更高层次的加工阶段，因听觉经验而产生格式塔联想。 Although the two stages work independently of each other, they always determine the properties of 虽然这两个阶段的工作相互独立，但它们始终决定着
one or multiple simultaneous auditory events in a conjoint manner. The rigorous differentiation of these two stimulus evaluation stages corresponds entirely to the two elementary areas of auditory experience. 一个或多个同时发生的听觉事件。这两个刺激评估阶段的严格区分完全符合听觉经验的两个基本领域。 The received ear signals can be attributed to the two sound source characteristics of "location" and "signal", which are independent of each other but always occur in a pair-wise fashion. 接收到的耳朵信号可归因于 "位置 "和 "信号 "这两个声源特征，它们相互独立，但总是成对出现。 Therefore, the presented association model is in agreement with many phenomena related to localisation in the superimposed sound field......" 因此，提出的关联模型与叠加声场中与定位有关的许多现象相吻合......"
12. Confirmation of a need to process timing information: 12.确认需要处理时间信息：
Source: 资料来源
Gave the following summary: 总结如下
"..The time-frequency uncertainty principle states that the product of the temporal and frequency extents of a signal cannot be smaller than . We study human ability to simultaneously judge the frequency and the timing of a sound. Our subjects often exceeded the uncertainty limit, sometimes by more than tenfold, mostly through remarkable timing acuity. "......时频不确定性原理指出，信号的时间和频率范围的乘积不能小于 。我们研究了人类同时判断声音频率和时间的能力。我们的受试者经常超过不确定性极限，有时甚至超过十倍，这主要是由于他们对时间的敏锐度非常高。 Our results establish a lower bound for the nonlinearity and complexity of the algorithms employed by our brains in parsing transient sounds, rule out simple "linear filter" models of early auditory processing, and highlight timing acuity as a central feature in auditory object processing...." 我们的研究结果为我们大脑在解析瞬态声音时所采用算法的非线性和复杂性确定了一个下限，排除了早期听觉处理的简单 "线性滤波器 "模型，并强调了时间敏锐性是听觉对象处理的一个核心特征...."
And further: 还有
"...In many applications such as speech recognition or audio compression (e.g. MP3 [18]), the first computational stage consists of generating from the source sound sonogram snippets, which become the input to latter stages. "......在语音识别或音频压缩（如 MP3 [18]）等许多应用中，第一个计算阶段包括从源声音中生成声波图片段，这些片段成为后面阶段的输入。 Our data suggest this is not a faithful description of early steps in auditory transduction and processing, which appear to preserve much more accurate information about the timing and phase of sound components than about their intensity..." 我们的数据表明，这并不是对听觉转导和处理早期步骤的忠实描述，听觉转导和处理似乎保留了关于声音成分 的时间和相位的准确信息，而不是关于其强度的准确信息......"
And finally: 最后
"...Early last century a number of auditory phenomena, such as residue pitch and missing fundamentals, started to indicate that the traditional view of the hearing process as a form of spectral analysis had to be revised. "......上世纪初，一些听觉现象，如残余音高和基音缺失等，开始表明必须对听觉过程作为一种频谱分析形式的传统观点进行修正。 In 1951, Licklider [25] set the foundation for the temporal theories of pitch perception, in which the detailed pattern of action potentials in the auditory nerve is used [26, 28], as opposed to spectral or place theories, in which the overall amplitude of the activity pattern is evaluated without detailed access to phase information. 1951 年，Licklider [25] 为音高感知的时间理论奠定了基础，在这种理论中，使用的是听觉神经中动作电位的详细模式[26, 28]，而不是频谱或位置理论，在频谱或位置理论中，评估的是活动模式的整体振幅，而没有详细的相位信息。 The groundbreaking work of Ronken [22] and Moore [23] found violations of uncertainty-like products and argued for them to be evidence in favour of temporal models. 朗肯[22]和摩尔[23]的开创性工作发现了类似不确定性产品的违规行为，并认为它们是支持时间模型的证据。 However this line of work was hampered fourfold, by lack of the formal foundation in time-frequency distributions we have today, by concentrating on frequency discrimination alone, by technical difficulties in the generation of the stimuli, and not the least by lack of understanding of cochlear dynamics, since the active cochlear processes had not yet been discovered. 然而，这项工作受到了四方面的阻碍：缺乏我们今天所拥有的时间-频率分布的正式基础；只专注于频率辨别；产生刺激的技术困难；尤其是缺乏对耳蜗动力学的了解，因为当时还没有发现耳蜗的主动过程。
Perhaps because of these reasons this groundbreaking work did not percolate into the community at large, and as a result most sound analysis and processing tools today continue to use models based on spectral theories. We believe it is time to revisit this issue....." 也许正是由于这些原因，这项开创性的工作并没有渗透到整个社区，因此今天大多数声音分析和处理工具仍在使用基于频谱理论的模型。我们认为，现在是重新审视这一问题的时候了....."
13. Transient and localization 13.瞬态和定位
Some very interesting information on transients and localization comes from the development work of Joseph Manger. The whole paper is recommended for reading. 约瑟夫-曼格（Joseph Manger）的开发工作提供了一些关于瞬态和本地化的非常有趣的信息。建议阅读整篇论文。
2/ Perception and hearing explained by new research 2/ 新研究解释了感知和听觉
The human hearing mechanism does not just detect the existence of sound, it also estimates the direction of the source as well as analysing the content of the sound to determine the most likely cause. In musical sounds, the pitch will also be determined. 人类的听觉机制不仅能检测声音的存在，还能估计声音来源的方向，并分析声音的内容，以确定最可能的原因。在音乐声中，音高也会被确定。 Josef Manger has been studying these mechanisms for over 20 years. He has found that each mechanism takes a different time to operate following an initial transient. The location and nature of the sound source are completely discerned before the pitch is recognised. 约瑟夫-曼格研究这些机制已有 20 多年。他发现，每种机制在初始瞬态后的运行时间都不同。在识别音高之前，声源的位置和性质已被完全辨别出来。
Pitch and timbral recognition is described by the well-established place theory, described in Part I, in which different parts of the basilar membrane resonate according to the frequencies in the sound. 基底膜的不同部位会根据声音的频率产生共鸣。 However, various authorities, such as Keidel, Spreng, Klinke and Zenner, have suggested that there is another, faster acting, mechanism which works in the time domain. 然而，Keidel、Spreng、Klinke 和 Zenner 等多位权威人士认为，在时域中还有另一种作用更快的机制。
The theory could not be tested with conventional loudspeakers. Confirmation of the theory was not possible until Josef Manger used his newly developed transducer as the sound source. 该理论无法用传统扬声器进行测试。直到约瑟夫-曼格使用他新开发的换能器作为声源，这一理论才得以证实。
Fig. 3 illustrates this principle of transient analysis and shows an idealised transient pressure waveform following an acoustic event. There are three important points made in the figure: 图 3 演示了这一瞬态分析原理，并显示了声学事件发生后的理想瞬态压力波形。图中有三个要点：
Fig. 3 图 3
I/ A complete cycle is quite unnecessary for the recognition of the sound source. Only the initial transient pressure change is required. The time of arrival of the transient at the two ears will be different and will locate cause, i.e. the source laterally within around a millisecond. I/ 识别声源完全不需要一个完整的周期。只需要最初的瞬态压力变化 。瞬态压力到达两只耳朵的时间会有所不同，并会在大约一毫秒内找到原因，即声源的横向位置。
2/ Following the event which generated the transient, the air pressure equalises itself along the line B-F. The period of time between B and F varies and allows the listener to establish the likely size of the sound source. 2/ 在产生瞬态的事件发生后，气压沿 B-F 线自行平衡。B 和 F 之间的时间段各不相同，听众可据此确定声源的可能大小。
3/ Only after the recognition of the source from the transient is the pitch recognised according to the place theory of the basilar membrane from the part of the wave-form beyond . 3/ 只有从瞬态中识别出声源后，才能根据基底膜的位置理论，从 以外的波形部分识别出音高。
The information in the initial transient pressure waveform goes beyond locating the source. Fig. 4 illustrates how the size of a sound source affects the pressure equalisation time. Pressure waveforms from a hand gun, a rifle and a cannon are shown. 初始瞬态压力波形中的信息不仅限于声源定位。图 4 说明声源的大小如何影响压力均衡时间。图中显示了手枪、步枪和大炮的压力波形。 It will be seen that the larger the source, the longer the pressure equalisation time. 可以看出，源越大，压力平衡时间就越长。
Fig. 4 图 4
How distorted transients can be - Manger illustrates it on the following pictures: 瞬态失真有多严重 - Manger 在下面的图片中进行了说明：
closed to the ideal 封闭于理想
14. AES Technical Document on phase accuracy and transient fidelity. 14.AES 关于相位精度和瞬态保真度的技术文件。
In 2002, the AESTD1001.1.01-10 drove the stake in the ground, and pegged the 10 usec as the maximum allowed timing difference between stereo loudspeakers across the entire audio band. 2002 年，AESTD1001.1.01-10 将 10 usec 定义为整个音频频段内立体声扬声器之间允许的最大时差。
Table 3. Suggestions for reference monitor loudspeakers and advice for home loudspeakers. 表 3.对参考监听扬声器的建议和对家用扬声器的建议。
Parameters 参数
Units/Conditions 单位/条件
Value 价值
Amplitude/frequency response 振幅/频率响应
Tolerance 4 dB 容差 4 分贝
偏离
偏离
Deviation to
Deviation to
前部 之间的差异
扬声器
Difference between front
loudspeakers
在范围内
至 2 kHz
In the range
to 2 kHz
0.5 dB 0.5 分贝
Directivity index 指向性指数
非线性失真衰减
)
Nonlinear distortion attenuation
)
临时保真度
衰减时间 用于还原为。
级别
, 即、0.37 的输出。
level
ransient fidelity
Decay time for reduction to a
level of , i.e., 0.37 of output
level
（最好 )
(preferably )
Time delay
立体声之间的区别
扬声器
Time delay
Difference between stereo
loudspeakers
系统动态范围
最高运行级别
（测量依据 IEC 60268，§
17.2, 指 1 米距离）
pere
System dynamic range
Maximum operating level
(measurement acc. to IEC 60268, §
17.2, referred to 1 m distance)
pere
（在 IEC 60268 程序仿真时） 噪声或特殊条件）
(at IEC 60268 program simulation
noise or special condition)
Noise level 噪音水平
15. Audibility of transients 15.瞬态的可听度
I have come across an interesting paper in JAES ,Vol.38, No.11,1990 November, "On the Correlation between the Subjective Evaluation of Sound and the Objective Evaluation of Acoustic Parameters for a Selected Source". 我在 1990 年 11 月出版的 JAES 第 38 卷第 11 期上看到一篇有趣的论文，题为 "关于选定声源的声音主观评价与声学参数客观评价之间的相关性"。
The authors performed subjective and objective analysis of several woofers using impulsive tones, and concluded: 作者利用脉冲音对几个低音扬声器进行了主观和客观分析，并得出结论：
"...A detailed analysis of the results of the subjective evaluation of loudspeakers showed that the subjective evaluation of the obtained sounds was decisively influenced by the work of the loudspeaker in a transient state. "......对扬声器主观评价结果的详细分析显示，对所获得声音的主观评价受到扬声器瞬态工作的决定性影响。 It appeared that the longer the duration of final transient and the smaller the value of coefficient , the greater the sharpness of the sounds emitted by the loudspeaker. 最终瞬态的持续时间越长，系数 的值越小，扬声器发出的声音就越尖锐。
16. Transients and Localization 16.瞬态和定位
The following paper, clearly indicates, that transients are critical in localization process. 以下论文清楚地表明，瞬态在定位过程中至关重要。
I have located an interesting piece of information in the paper "Localization of sound in rooms", from JASocAm. 74 (5) Nov 1983. The paper is by WM Hartman from Michigan State University, Dept. of Physics, and provides the following summary: 我在 JASocAm 的论文 "室内声音定位 "中找到了一条有趣的信息。74 (5) Nov 1983。论文作者是密歇根州立大学物理系的 WM-哈特曼，摘要如下：
"...This paper is concerned with the localization of sources of sounds by human listeners in rooms. It presents the results of source-identification experiments designed to determine whether the ability to localize sound in a room depends upon the room acoustics, and how it depends upon the nature of the source signal. "......本文涉及人类听者在室内对声源的定位。本文介绍了声源识别实验的结果，这些实验旨在确定在房间中定位声音的能力是否取决于房间的声学条件，以及如何取决于声源信号的性质。
The experiments indicate that the localization of impulsive sounds, with strong attack transients, is independent of the room reverberation time, though it may depend upon the room geometry. 实验表明，具有强攻击瞬态的脉冲声的定位与房间混响时间无关，但可能取决于房间的几何形状。
For sounds without attack transients, localization improves monotonically with the spectral density of the source. 对于没有攻击瞬态的声音，定位效果随着声源频谱密度的增加而单调增强。
Localization of continuous broadband noise does depend upon room reverberation 连续宽带噪声的定位确实取决于室内混响
time .." 时间......"
More papers by Hartmann and Rakerd. 哈特曼和瑞克德的更多论文。
"Localization of sound in rooms, II: The effects of a single reflecting surface" "室内声音定位，II：单一反射面的影响
"....Our results indicate the following: (1) A sound must include transients if the precedence effect is to operate as an aid to its localization in rooms. (2) Even if transients are present the precedence effect does not eliminate all influences of room reflections. "....，我们的研究结果表明了以下几点：(1)如果先声效果要帮助声音在房间中定位，声音必须包括瞬态。 (2)即使存在瞬态，先声效果也不能消除房间反射的所有影响。 (3) Due to the interference of reflections large interaural intensity differences may occur in a room and these have a considerable influence on localization; this is true even at low frequencies for which IID cues do not exist in a free field. (3) 由于反射的干扰，房间里可能会出现很大的耳际强度差异，这对定位有相当大的影响；即使在低频时也是如此，因为在自由场中不存在 IID 线索。 (4) Listeners appear to have certain expectations about the reliability and plausibility of various directional cues and perceptually weight the cues accordingly; we suggest that this may explain, in part, the large variation in time-intensity trading ratios reported in the literature and also the differing reports regarding the importance of onsets for localization. (4) 听者似乎对各种方向性线索的可靠性和可信度有一定的预期，并据此对线索进行知觉加权；我们认为，这可能部分解释了文献中报道的时间强度交易比率的巨大差异，以及关于起始点对定位的重要性的不同报道。 (5) In this study we find that onset cues are of some importance to localization even in free field. (5) 在这项研究中，我们发现即使在自由场中，起始线索对定位也有一定的重要性。
"Localization of sound in rooms: III: Onset and duration effects" "室内声音定位：III：起始和持续时间效应
"...(1) A rapid onset facilitates localization in a free field by a measurable but small amount, about 0.5 deg . "......(1)快速启动可促进自由场中的定位，其幅度可测量但很小，约为 0.5 度。 It facilitates localization in rooms by substantially larger amount because the onset allows the precedence effect to operate and without the precedence effect localization is poor due to misdirection cues in steady-state sound field. 它能大大促进房间内的定位，因为起始音允许先声效应发挥作用，而如果没有先声效应，由于稳态声场中的误导线索，定位效果会很差。
(2) The precedence effect is maximally effective when the signal onset is instantaneous. Its effectiveness begins to diminish as the onset duration is increased....." (2) 当信号瞬时发生时，优先效应的效果最大。随着起始时间的延长，其效果开始减弱....."
17. More on localization and transients 17.关于定位和瞬态的更多信息
In a paper by Brad Rakerd and William M. Hartmann "Localization of noise in a reverberant environment" (Michigan State University), they conclude: Brad Rakerd 和 William M. Hartmann 在一篇题为 "混响环境中的噪声定位"（密歇根州立大学）的论文中得出结论："在混响环境中的噪声定位"：
"...(1) Localization of noise is enhanced by an attack transient. An attack transient appears to be particularly helpful when the direct-reverberant ratio is low. Attack transients give an advantage over slow onsets when the reflections are not much delayed re the direct sound. "......(1)攻击瞬态可加强噪声的定位。当直达声与反射声的比值较低时，攻击瞬态似乎特别有用。当反射声与直达声之间的延迟不大时，攻击瞬态比慢速瞬态更有优势。 By contrast, attack transients are of only marginal value 相比之下，攻击瞬态的价值微乎其微
when noise is presented by headphones or tones are presented in an anechoic room (Tobias and Schubert, 1959; Rakerd and Hartmann, 1986). 当噪音通过耳机或音调在消声室中呈现时（Tobias 和 Schubert，1959 年；Rakerd 和 Hartmann，1986 年）。
(2) Onsets are a great leveler among individuals. Whereas the ability to localize steady steady-state sounds varies greatly among listeners, the ability to localize sounds with an onset transient shows best to worst differences less than 1.5 degrees among our seven listeners...." (2) 起音在个体间有很大的均衡性。不同听者对稳定的稳态声音的定位能力差异很大，而对起始瞬态声音的定位能力，在我们的七位听者中，从最好到最差的差别不到 1.5 度...."。
18. Even more on localization and transients 18.更多关于本地化和瞬态的内容
"Localization of sound in a room with reflecting walls" W.M. Wagenaars "有反射墙壁的房间内的声音定位" W.M. Wagenaars
"....3. CONCLUSIONS "....3.结论
In this study localization of sound in a room with reflecting walls was tested. Eleven stimuli were used, differing in spectral and temporal information. For such a room the following can be concluded: 这项研究测试了在有反射墙壁的房间中声音的定位。共使用了 11 种不同频谱和时间信息的刺激。对于这样的房间，可以得出以下结论：
Signal bandwidth is an important cue for localization. The broader the frequency spectrum of a sound, the better localization performance. 信号带宽是定位的一个重要提示。声音的频谱越宽，定位性能就越好。
Offsets seem to be an equally important cue for localization as onsets. Localization performance are similar for signals with an abrupt onset, offset, or both. 偏移似乎是与起始信号同样重要的定位线索。对于突发性信号、偏移信号或两者都有的信号，定位效果相似。
Localization performance for steady state sinusoids is frequency-dependent. For simply gated sinusoids performance is not dependent of frequency. 稳态正弦波的定位性能与频率有关。对于简单选通的正弦波，其性能与频率无关。
Although many of the errors made were distance errors, subjects are able to localize distance quite well. Furthermore subjects usually select the correct side, even for the hard to localize steady state sinusoids....." 虽然许多错误都是距离错误，但受试者能够很好地定位距离。此外，受试者通常会选择正确的一侧，即使是难以定位的稳态正弦波....."
19. Sound Quality and Transient Response. 19.音质和瞬态响应
In the next paper: "Correlation of Transient Measurements on Loudspeakers with Listening Tests" by M. Corrington, published in JAES, JANUARY 1955, VOLUME 3, NUMBER 1, we find an interesting measurement method, allowing for separation of the "overhang transient" - see below 下一篇论文"在 M. Corrington 发表于 1955 年 1 月 JAES 第 3 卷第 1 期的《扬声器瞬态测量与听力测试的相关性》一文中，我们发现了一种有趣的测量方法，可以将 "悬空瞬态 "分离出来 - 参见下文
The paper reads well, and has the following interesting conclusion: 这篇论文读起来朗朗上口，并有如下有趣的结论：
"....This information supplements the steady-state sound pressure measurements. "....，这些信息是对稳态声压测量结果的补充。 We have never found any system with low transient distortion that did not also have a smooth sound-pressure curve; on the other hand, we have measured systems with fairly sharp and small peaks in the sound-pressure response that produced objectionable transient distortion. 我们从未发现过瞬态失真低但声压曲线不平滑的系统；另一方面，我们也曾测量过声压响应峰值相当尖锐和微小的系统，但却产生了令人反感的瞬态失真。
There is very good correlation between transient distortion and subjective listening tests. Whenever there are peaks in the transient distortion, one can be sure that the listening tests will reveal unpleasant distortion, even though the sound-pressure curve is quite smooth.... 瞬态失真与主观听音测试之间有很好的相关性。只要瞬态失真出现峰值，就可以肯定听力测试会出现令人不悦的失真，即使声压曲线非常平滑....。
Extensive measurements show that for a high-quality audio system the sound-pressure curve must be smooth and properly shaped, and that the transient distortion should be down at least 18 dB throughout the range. 大量测量结果表明，对于高品质的音频系统而言，声压曲线必须平滑且形状正确，整个音域的瞬态失真至少应降低 18 分贝。 One can then be fairly certain that the system will pass very careful listening tests...." 这样，我们就可以相当肯定，该系统将通过非常仔细的听力测试...."
20. Confirmation of two-stage processing by the ear, as discussed in (3) and (11). 20.如（3）和（11）所述，确认耳朵的两阶段处理。
Yet another interesting paper. It puts the early reflections in somewhat different perspective. 又是一篇有趣的论文。它从另一个角度阐述了早期的反思。
"The Significance of Early High-Frequency Reflections from Loudspeakers in Listening Rooms", Preprint 4094, David Moulton, David Moulton Professional Services, Groton, MA "听音室扬声器早期高频反射的意义"，4094 号预印本，David Moulton，David Moulton 专业服务公司，马萨诸塞州格罗顿市
"...Any reverberant space yields comb-filtering effects, and virtually all listening to music via loudspeakers is done in such spaces. "......任何混响空间都会产生梳状滤波效果，几乎所有通过扬声器聆听音乐的过程都是在这样的空间中完成的。 Therefore, logically speaking, all listening is done under compromised conditions, where a primary attribute of accurate sound reproduction (fiat amplitude response) is negated. 因此，从逻辑上讲，所有的聆听都是在受到损害的条件下进行的，在这种条件下，准确声音重现的一个主要属性（非标准振幅响应）被否定了。 Yet we must acknowledge that music playback systems seem to work well: listeners enjoy listening, they readily and accurately identify sounds (and will testify to their realism), and some listeners are able to detect truly microscopic differences between alternate components in the playback system. 然而，我们必须承认，音乐播放系统似乎运行良好：听众喜欢听，他们能轻易而准确地辨别声音（并能证明声音的真实性），有些听众还能发现播放系统中不同组件之间真正的微小差异。
This anomaly raises the question: how can individuals listen effectively to loudspeakers in reverberant spaces and why don't the ubiquitous comb-filtering interference effects always pose problems for the listener? 这种反常现象提出了一个问题：人如何才能在混响空间中有效地聆听扬声器的声音，为什么无处不在的梳状滤波干扰效应并不总是给聆听者带来问题？
I suggest that the answer lies in the nature of our auditory localization capability, which makes use of interference effects such as comb-filtering as a function of performing the sound source localization task. 我认为，答案就在于我们的听觉定位能力的本质，这种能力在执行声源定位任务时会利用梳状滤波等干扰效应。
That task is performed at a pre-conscious neurological stage and most early reflections are localization information that is not presented to the conscious mind. 这项任务是在前意识神经阶段完成的，大多数早期反映都是没有呈现在意识中的定位信息。 Further, we do not consciously perceive the amplitude response characteristics of comb-filtering effects that occur in reverberant spaces as a result of early reflections, even though such effects are clearly measurable...." 此外，我们不会有意识地感知到混响空间中由于早期反射而产生的梳状滤波效应的振幅响应特性，尽管这种效应显然是可以测量的...."
The above statement confirms earlier findings of Gunter Theile, Watkinson and Manger about the ear processing the incoming audio stimulus in two stages: The received ear signals can be attributed to the two sound source characteristics of "location" and "signal", which are independent of each other but always occur in a pair-wise fashion. 上述说法证实了冈特-泰尔（Gunter Theile）、沃特金森（Watkinson）和曼格（Manger）早先关于耳朵分两个阶段处理传入音频刺激的发现：耳朵接收到的信号可归因于 "位置 "和 "信号 "这两个声源特征，它们相互独立，但总是成对出现。
20. General Conclusions From Papers Presented Above 20.从上述论文中得出的一般性结论
First of all - the room itself. 首先是房间本身。
Accordingly to Bernd Theiss, Malcolm O. J. Hawksford in AES Preprint 4462: 根据 Bernd Theiss、Malcolm O. J. Hawksford 在 AES Preprint 4462 中的说法：
"...Early reflections . "......早期反思 。
Early reflections occurring less than 2.5 ms after the original sound sensation are known to shift the image towards their direction and to blur the image. 众所周知，在原始声音感觉后不到 2.5 毫秒发生的早期反射会使图像向其方向偏移并模糊图像。
Early reflections . 早期反思 。
Early reflections occurring more than 2.5 ms but less than 5 ms after the original sound sensation are known to blur the image, although they keep the direction of the image constant...." 众所周知，在原始声音感觉之后超过 2.5 毫秒但小于 5 毫秒的早期反射会模糊图像，尽管它们会保持图像的方向不变...."
So if your goal is to deliver the sharpest image, or most accurate localization, you would be well advised to take care of transient origination (loudspeakers) and also provide some acoustical treatment to the walls/room. 因此，如果您的目标是提供最清晰的图像或最准确的定位，建议您最好对瞬态源（扬声器）进行处理，并对墙壁/房间进行一些声学处理。
There are basically three areas where linear-phase loudspeakers differ from minimumphase loudspeakers. 线性相位扬声器与最小相位扬声器的区别主要体现在三个方面。
Linear-phase speakers provide more accurate spatial information, rather than timbral. Tonal balance is the same for both loudspeaker types. 线性相位扬声器能提供更准确的空间信息，而不是音色。两种扬声器的音调平衡是相同的。 This is where the tests are falling apart, because listeners are looking for tonal differences, rather than subtle spatial clues - sharper image, better located soloists, stage depth. It's subtle, but it is there. 这正是测试失败的原因所在，因为听众寻找的是音调差异，而不是微妙的空间线索--更清晰的图像、独奏者更好的位置、舞台深度。这很微妙，但确实存在。
Identical phase response for all loudspeaker in the system. The phase response in correctly equalized multi-channel linear-phase system is 0 deg in every loudspeaker. Therefore it immediately satisfies AESTD1001.1.01-10 for phase accuracy and transient fidelity to perfection. 系统中所有扬声器的相位响应相同。在正确均衡的多声道线性相位系统中，每个扬声器的相位响应均为 0 度。因此，它能立即满足 AESTD1001.1.01-10 对相位精度和瞬态保真度的完美要求。 The measurements of linearphase loudspeaker are presented on my website, and comments on AESTD1001.1.01-10 are presented in http://www.bodziosoftware.com.au/AES_Document_Comments.pdf 线性相位扬声器的测量结果见我的网站，对 AESTD1001.1.01-10 的评论见 http://www.bodziosoftware.com.au/AES_Document_Comments.pdf 。
Another interesting paper from th AES Convention. I would recommend reading the entire paper. AES 大会上的另一篇有趣论文。我建议您阅读整篇论文。
Source: "Directions for Qualified Loudspeaker Evaluations", AES Preprint 3603, Peter M. Pfleiderer, 1993. 资料来源"合格扬声器评估指南》，AES 预印本 3603，Peter M. Pfleiderer，1993 年。
The paper concludes with the following summary: 本文最后总结如下：
"...Summary "......摘要
An almost unbelievable state of perfection has been reached for electronic components within the electroacoustical reproduction chain due to competent applications of measurement technology. 由于测量技术的有效应用，电声再现链中的电子元件已经达到了几乎令人难以置信的完美状态。 With loudspeakers, on the other hand, competent measurement methods are currently not even in practical use. Obviously, test methods are required which are capable of uncovering major changes to signal waveforms originating from linear and acoustical errors. 另一方面，对于扬声器，目前还没有实用的测量方法。显然，我们需要的测试方法必须能够揭示因线性误差和声学误差引起的信号波形的重大变化。
Measurements with square wave signals should be included as standard testing procedures in order to be able to detect errors with sound quality and spatial imaging in all HiFi components, but especially in loudspeaker systems. Many technical and acoustical faults can namely not be registered with SPL or frequency measurements, although they have induced significant irregularities into the relevant audio signal waveform. 方波信号测量应作为标准测试程序，以便能够检测所有高保真组件，尤其是扬声器系统的音质和空间成像错误。许多技术和声学故障无法通过声压级或频率测量记录下来，尽管这些故障会导致相关音频信号波形出现明显的不规则。
This is the reason why loudspeakers of proven square wave response capability are an important prerequisite for the natural reproduction of sound. Moreover, it is only possible to detect acoustic faults with this type of technically faultless reference loudspeaker. 正因为如此，具有经过验证的方波响应能力的扬声器是自然再现声音的重要先决条件。此外，只有使用这种技术上无故障的参考扬声器，才有可能检测出声学故障。 It should be clearly noted that all other current components in the electroacoustical reproduction chain already transmit square wave signals correctly. 需要明确指出的是，目前电声再现链中的所有其他组件都已能正确传输方波信号。
Correct square wave reproduction with loudspeakers has the same importance as was the case for correct square wave reproduction with amplifiers in the 1960's. Both constitute fundamental advances and establish important conditions for high quality reproduction of music. 扬声器正确再现方波的重要性与 20 世纪 60 年代放大器正确再现方波的重要性相同。两者都是根本性的进步，并为高质量的音乐再现创造了重要条件。 Nothing can propagate the concept of high fidelity more than these types of advances...." 没有什么比这类预付款更能宣传高保真概念了...."。
Time Domain Instrument Testing 时域仪器测试
Real-life loudspeaker example 实际扬声器示例
The system under test discussed here consists of a filter and a loudspeaker in an enclosure. These two components that will introduce time delay are the filter and the combination of driver and the enclosure itself. 这里讨论的测试系统包括一个滤波器和一个外壳中的扬声器。这两个会产生时间延迟的部件是滤波器以及驱动器和箱体本身的组合。 To illustrate the above, a 12 " guitar loudspeaker in a vented box was measured and it's minimum-phase responses were obtained with a help of an MLS measurement technique - see below. It is immediately observable, that the loudspeaker has rather irregular frequency response. 为了说明上述问题，我们测量了一个通风箱中的 12 英寸吉他扬声器，并利用 MLS 测量技术获得了其最小相位响应（见下文）。可以立即看出，该扬声器的频率响应相当不规则。 Since the loudspeaker is essentially a minimum-phase device, the corresponding phase response is also highly irregular, and definitely not flat. 由于扬声器本质上是一种最小相位设备，因此相应的相位响应也极不规则，而且绝对不平坦。
Let's establish the frequency response of interest, which is the frequency range where the SPL will be equalized to flat response. In my example it will be: . 让我们确定感兴趣的频率响应，也就是将声压级均衡为平坦响应的频率范围。在我的例子中，它将是 。
A 300 Hz square wave reproduced by this loudspeaker is highly distorted. Strong ringing is due to 10 dB sharp SPL peak located at 3.5 kHz . You can see, that there are about 11 periods of ringing waveform in one period of 300 Hz square wave. 该扬声器再现的 300 赫兹方波失真度很高。位于 3.5 千赫处的 10 分贝尖锐声压级峰值产生了强烈的振铃。您可以看到，在一个周期的 300 赫兹方波中，大约有 11 个周期的振铃波形。
Instrument test results obtained from linear-phase loudspeakers reveal their true superiority in time domain. The following test results were obtained by John Kreskovsky of Music and Design ( http://www.musicanddesign.com ) 线性相位扬声器的乐器测试结果显示了其在时域方面的真正优势。以下是 Music and Design 公司的 John Kreskovsky 获得的测试结果 ( http://www.musicanddesign.com )
As John points out: "....The measurements were not taken in an anechoic environment and are of the continuous time type, recorded over numerous cycles, windowing over a reflection free period can not be performed. 约翰指出"....The 测量不是在消声环境中进行的，而且是连续时间类型的测量，记录了无数个周期，因此无法在无反射期间进行窗口测量。 Thus, there is some contamination by room reflections resulting is some degradation in the observed response. 因此，房间反射会造成一些污染，导致观测到的响应有所降低。
The first figure shows the 300 Hz response. 第一张图显示的是 300 赫兹的响应。 This is close to the low frequency cut off of the system where the phase rotation and group delay due to the 200 Hz high pass cut off would normally result in loss of flat top behaviour and the 2 k Hz crossover would cause distortion of the initial rise. 这接近于系统的低频截止点，在这里，200 赫兹高通截止点引起的相位旋转和群延迟通常会导致平顶特性的丧失，而 2 k 赫兹的分频器则会导致初始上升失真。 This is shown in the insert at the upper right of the plot for the linearized system and confirmed by the lower plot which if for the standard LR4 system. The white trace is the input, orange the acoustic output from the speaker system. 线性化系统的右上方插入图显示了这一点，而标准 LR4 系统的下部插入图也证实了这一点。白色轨迹为输入，橙色轨迹为扬声器系统的声音输出。
300 Hz square response of linearized system, left, and standard LR4 crossover, right. 左为线性化系统的 300 赫兹方波响应，右为标准 LR4 分频器的 300 赫兹方波响应。
500 Hz square response of linearized system, left, and standard LR4 crossover, right. 左侧为线性化系统的 500 赫兹方波响应，右侧为标准 LR4 分频器的 500 赫兹方波响应。
1 kHz square response of linearized system, left, and standard LR4 crossover, right. 左为线性化系统的 1 kHz 方波响应，右为标准 LR4 分频器的 1 kHz 方波响应。
2 kHz square response of linearized system, left, and standard LR4 crossover, right. 左为线性化系统的 2 kHz 方波响应，右为标准 LR4 分频器的 2 kHz 方波响应。
...." End of quote. ...."引用完毕。
My own measurements on 18 " McCauley subwoofers further confirm time domain superiority of linear-phase loudspeakers. 我自己对 18 英寸 McCauley 低音炮的测量结果进一步证实了线性相位扬声器在时域上的优越性。
20Hz square wave: Linear-Phase Mode and Minimum-Phase Mode 20Hz 方波：线性相位模式和最小相位模式
Shown above, the time-domain comparison measurement results speak for themselves. It needs to be remembered, that we are dealing here with a very heavy- 如上图所示，时域对比测量结果不言自明。需要记住的是，我们在这里处理的是一个非常重的--......。
coned, 18" driver, low-pass filtered, in a vented (resonating) enclosure, and yet, the time domain performance is near-perfect accurate. It's pretty amazing to see a vented loudspeaker, holding the acoustic pressure nearly constant for 25 ms . 在一个通风（共振）箱体内的 18 英寸锥形驱动器，经过低通滤波，时域性能却近乎完美精确。看到一个通风扬声器在 25 毫秒内保持声压几乎恒定，真是令人惊叹。
Next, I used 2ms-wide pulses separated by 350 ms space as the source signal. On the 2 ms pulse, the minimum-phase version delivered a more of a "thump" instead of a pop or a click. 接下来，我使用 2 毫秒宽、间隔 350 毫秒的脉冲作为信号源。在 2 毫秒的脉冲中，最小相位版本发出的更多是 "砰 "的一声，而不是 "啪 "或 "咔嗒 "声。 This is perhaps not surprising, as the post-ringing of the pulse extended to 130 ms and far exceeded the 30 ms "memory effect" of the auditory system. Here, the driver, filter and vented enclosure added it's own, combined signature. 这也许并不奇怪，因为脉冲响后时间延长至 130 毫秒，远远超过了听觉系统 30 毫秒的 "记忆效应"。在这里，驱动器、滤波器和通风罩共同增添了自己的特征。 It is also observable, that the minimum-phase version of the subwoofer has converted the clearly asymmetrical pulse into a much more symmetrical bi-polar pulse with post-ringing. This is clearly visible on the screen shots below. 还可以看到，最小相位版本的低音炮已将明显不对称的脉冲转换为更为对称的双极性脉冲，并伴有后振铃。这在下面的屏幕截图中清晰可见。
5ms Impulse in Linear-Phase Mode and Minimum-Phase Mode 线性相位模式和最小相位模式下的 5ms 脉冲
When a 2 ms bi-polar pulse was used for excitation, the minimum-phase version has done the opposite, and converted the symmetrical bi-polar pulse into a pulse with clear asymmetrical tendency. 当使用 2 毫秒的双极性脉冲进行激励时，最小相位版本却反其道而行之，将对称的双极性脉冲转换为具有明显不对称趋势的脉冲。 The ringing past the pulse is due to a more distant microphone placement, so now, the mike picks some of the room reflections. 脉冲过后的振铃是由于麦克风放置的位置更远了，因此麦克风现在能接收到一些房间反射。
2ms Bi-polar pulse in Linear-Phase Mode and Minimum-Phase Mode 线性相位模式和最小相位模式下的 2ms 双极性脉冲
When a 10 ms bi-polar pulse was used for excitation, the minimum-phase version has even more asymmetrical tendency. 当使用 10 毫秒双极性脉冲进行激励时，最小相位版本的不对称趋势更大。
10ms Bi-polar pulse in Linear-Phase Mode and Minimum-Phase Mode 线性相位模式和最小相位模式下的 10ms 双极性脉冲
Finally some more square wave measurements from UE User's Manual. 最后是 UE 用户手册中的方波测量结果。
The linear phase result is on the left and the nonlinear phase result on the right. It should be noted that there is some distortion in the wave forms that that must be attributed to room reflections. 左边是线性相位结果，右边是非线性相位结果。值得注意的是，波形中存在一些失真，这必须归咎于房间反射。 Square wave testing is a steady state test and without a true anechoic chamber the effects of room reflections can not be eliminated. 方波测试是一种稳态测试，如果没有真正的消声室，就无法消除房间反射的影响。
Never the less, for the 300 Hz case shown in the first figure, the linear phase system shows the sharp rise and fairly flat top expected. 尽管如此，对于第一幅图中显示的 300 赫兹情况，线性相位系统显示出了预期的急剧上升和相当平缓的顶部。 The nonlinear phase case shows early tweeter response followed by the woofer response and the sloped top is an artefact of the nonlinear phase. The response also significantly overshoots the correct level. 在非线性相位情况下，高音扬声器的早期响应紧随低音扬声器的响应之后，倾斜的顶部是非线性相位的假象。响应也明显超出了正确的电平。 This latter effect is seldom discussed when comparisons of linear and nonlinear phase systems are made. 在对线性和非线性相位系统进行比较时，很少讨论后一种效应。 Even though the amplitude of reach frequency component is correctly reproduced in the nonlinear phase system, the lack of linear phase means that the different frequency components do not sum correctly since that are delayed by different amounts. 尽管非线性相位系统能正确再现频率分量的振幅，但由于缺乏线性相位，不同频率分量的总和并不正确，因为它们的延迟量不同。 The overshoot is a result of time distortion. 过冲是时间失真的结果。
300 Hz square wave response, Linear phase, left; Nonlinear phase, right 300 赫兹方波响应，线性相位，左侧；非线性相位，右侧
The next figure shows the same comparison for a 1 kHz square wave. Again, some distortion is observed due to room reflections. However, the linear phase case 下图显示了 1 kHz 方波的相同对比。同样，由于房间反射的原因，也出现了一些失真。然而，线性相位情况
again shows the expected sharp rise and relatively flat top. The nonlinear phase system more clearly shows the time lag between the woofer and tweeter response. 再次显示出预期的急剧上升和相对平坦的顶部。非线性相位系统更清晰地显示了低音扬声器和高音扬声器响应之间的时滞。
The next figure shows the result for a 3 k Hz square wave. The differences between linear and nonlinear phase, while clearly evident, are less significant because the fundamental is above the crossover point and there is little contribution from the woofer due to the order low pass response. With the system designed another interesting feature of the linear phase system can be examined, the effect of crossover slope. 下图显示的是 3 k Hz 方波的结果。线性相位和非线性相位之间的差异虽然明显，但并不那么显著，因为基频在分频点之上，而且由于 阶低通滤波器的响应，低音扬声器的贡献很小。系统设计完成后，我们就可以研究线性相位系统的另一个有趣特性，即分频斜率的影响。
The next figure shows the 1 kHz response of the linear phase and nonlinear phase system when the slope of the woofer to tweeter crossover is increased to order, octave. With the Ultimate Equalizer this is easily accomplished by selecting the new octave slopes and clicking Show complete system to calculate and load the new filters. 下图显示了当低音扬声器到高音扬声器分频器的斜率增加到 阶 倍频程时，线性相位和非线性相位系统的 1 kHz 响应。使用终极均衡器，只需选择新的 倍频程斜率，并单击显示完整系统来计算和加载新的滤波器，即可轻松实现这一操作。
1 kHz response of linear and nonlinear phase system with order crossover. 具有 阶分频器的线性和非线性相位系统的 1 kHz 响应。
This result should be compared to that of figure where the crossover was order. Changing the order has no effect on the linear phase system at the design point. The nonlinear phase system response is significantly different solely due to the change in crossover order. 应将这一结果与分频阶数为 的图表进行比较。改变阶数对设计点的线性相位系统没有影响。而非线性相位系统的响应则完全由于分频阶数的改变而大不相同。
Finally, the last figure shows the effect of reducing the crossover to order. The response of the nonlinear phase system looks somewhat better now. However, for flat response the tweeter must be connected with inverted polarity in the nonlinear phase system and the initial tweeter pulse is therefore in the wrong direction. 最后，最后一张图显示了将分频减小到 阶的效果。现在，非线性相位系统的响应看起来有所改善。不过，为了获得平坦的响应，高音扬声器必须与非线性相位系统中的极性相反，因此高音扬声器的初始脉冲方向是错误的。 It should be noted that many audio enthusiasts feel the order crossover sound better than those of higher order. This may be a result of the improved wave form observed here and could be an indication of the potential of linear crossover and speakers of any order since they will all preserve wave form relative to the design point. 值得注意的是，许多音响发烧友认为 阶的分频器比更高阶的分频器音质更好。这可能是此处观察到的波形得到改善的结果，也可能是线性分频器和任何阶数扬声器潜力的体现，因为它们都能保持相对于设计点的波形。
1 kHz response of linear and nonlinear phase system with order crossover. 具有 阶分频器的线性和非线性相位系统的 1 kHz 响应。
Conclusions 结论
At the time of this writing, linear-phase loudspeakers are still a new "kid on the block". Past attempts in creating them resulted in offerings that were simply too expensive for wide-spread use. 在撰写本报告时，线性相位扬声器仍是 "新兴事物"。过去尝试制造的产品过于昂贵，无法广泛使用。 The most accurate implementation of linear-phase loudspeaker requires a full set of individual driver measurements, coupled with a DSP approach, in addition to an active amplification system. 要最精确地实现线性相位扬声器，除了有源放大系统外，还需要全套的单个驱动器测量，并结合 DSP 方法。 This really makes the linearphase system highly customized device - a world of difference in comparison to the current approach of loudspeaker industry. 这确实使线性相位系统成为高度定制化的设备，与目前扬声器行业的做法相比有着天壤之别。
However, this particular feature makes the linear-phase system an ideal DIY device. In our world, everything is custom-built, with an aim to typically outperform comparable commercial designs. 然而，这一特性使得线性相位系统成为理想的 DIY 设备。在我们的世界里，所有东西都是定制的，其目标通常是超越同类商业设计。 Linear-phase loudspeakers offer everything that minimum-phase loudspeakers can offer, and then reward you with often vastly superior performance in time domain, as explained in the pages above. 线性相位扬声器可提供最小相位扬声器所能提供的一切，而且在时域方面的表现往往更为出色，如上文所述。
It appears, that my poor and outdated listening/evaluating habits, coupled with lack of standard listening methodology for time/space-domain assessment of loudspeakers conspired to cloud my ability to really critically listen to the full set of my loudspeakers during some of my evaluation tests. 看来，我的听音/评估习惯很差而且过时，再加上缺乏对扬声器进行时域/空域评估的标准听音方法，这些因素共同影响了我在一些评估测试中真正批判性地聆听全套扬声器的能力。 Secondly, not every musical material will reveal all time-domain characteristics to the same degree. 其次，并非每种音乐素材都能在相同程度上显示出所有时域特征。 For instance, tight, well-defined bass, will manifest itself on gunshots and explosions in DVD movies, but will not stand out during low-frequency, seismic earthquake effects on LFE channel. 例如，紧凑、清晰的低音会在 DVD 电影的枪声和爆炸声中表现出来，但在 LFE 频道的低频地震效果中却不突出。 In more critical tests, I did pick the "tighter bass" characteristic, as it was too obvious to miss on the large, 18 " subs. Also, I pointed out earlier the effect of feeling closer to the orchestra, as if I could better discriminate their sitting arrangement. 在更关键的测试中，我确实发现了 "低音更紧凑 "的特点，因为这在 18 英寸的大型低音炮上太明显了，不可能错过。此外，我之前还指出了感觉更接近管弦乐队的效果，似乎我可以更好地分辨他们的坐姿安排。 Both of these effects have really nothing to do with frequency domain they are both more of the time/space domain phenomena. 这两种效应其实都与频域无关，它们更多的是时域/空域现象。
It is clear, that designing loudspeakers using frequency-domain characteristics as the main (or only) criteria leads to stagnated, oversimplified, and ultimately inaccurate system. 很明显，将频域特性作为主要（或唯一）标准来设计扬声器会导致系统停滞不前、过于简化，并最终导致系统不准确。 If I continued to design loudspeakers that never reveal timedomain or spatial-domain subtleties, I would never even know of the existence of such subtleties, therefore, I would never be motivated to change - thus allowing the vicious cycle to continue. 如果我继续设计从不显示时间域或空间域微妙之处的扬声器，我甚至永远不会知道这些微妙之处的存在，因此，我永远不会有改变的动力，从而使恶性循环继续下去。 It is evident, that the ear examines the incoming audio stimulus in two-stage process: (1) location - here the transient of the stimulus is examined, and (2) signal - here the spectral properties of the stimulus are examined. The two processes always work in-tandem. 很明显，耳朵会分两个阶段检查传入的音频刺激：(1) 位置--此处检查刺激的瞬态；(2) 信号--此处检查刺激的频谱特性。这两个过程始终同步进行。 It is therefore essential, that the loudspeaker provides undistorted waveforms to the auditory system to enable correct processing of both stages. 因此，扬声器必须向听觉系统提供不失真波形，以便正确处理这两个阶段。
So, here I am. Struggling to come out of the "frequency-domain box" and into the new world of time/frequency/space-domain characteristics of contemporary loudspeakers. But even at these early stages of adopting a new technology, I find it already very rewarding. 所以，我就来了。我正在努力摆脱 "频域 "的束缚，进入当代扬声器时域/频域/空域特性的新世界。但即使是在采用新技术的早期阶段，我也觉得收获颇丰。 This is because it's evident that a new, accurate and realistic acoustic transduction technology is being achieved in much more accessible commercial way. 这是因为很明显，一种新的、准确而逼真的声学传导技术正在以更容易获得的商业方式实现。
Lateral information (left, ahead, right) 横向信息（左、前、右）
For determining the lateral input direction (left, front, right) the auditory system analyzes the following ear signal information: 为了确定横向输入方向（左、前、右），听觉系统会分析以下耳朵信号信息：
Interaural time differences 耳内时差
Sound from the right side reaches the right ear earlier than the left ear. The auditory system evaluates interaural time differences from 来自右侧的声音比左耳更早到达右耳。听觉系统从以下方面评估耳间时差
o Phase delays at low frequencies o 低频时的相位延迟
0 group delays at high frequencies 0 高频群延迟
Interaural level differences 耳内电平差异
Sound from the right side has a higher level at the right ear than at the left ear, because the head shadows the left ear. These level differences are highly frequency dependent and they increase with increasing frequency. 来自右侧的声音在右耳的电平要高于左耳，因为头部会遮挡左耳。这些声级差异与频率高度相关，随着频率的增加而增大。
For frequencies below 800 Hz , mainly interaural time differences are evaluated (phase delays), for frequencies above 1600 Hz mainly interaural level differences are evaluated. Between 800 Hz and 1600 Hz there is a transition zone, where both mechanisms play a role. 对于低于 800 Hz 的频率，主要评估耳际时差（相位延迟），对于高于 1600 Hz 的频率，主要评估耳际电平差。在 800 Hz 和 1600 Hz 之间有一个过渡区，两种机制都在这里发挥作用。
Localization accuracy is degree for sources in front of the listener and 15 degrees for sources to the sides. Humans can discern interaural time differences of 10 microseconds or less. 听者前方声源的定位精度为 度，两侧声源的定位精度为 15 度。人类可以分辨出 10 微秒或更小的听觉时差。
