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| [#41] 改善音場深度和寬度 只用一對音箱的話，先看音響系統重播能力，再看房間在聲學上的影響。 音響系統重播能力可從三方面評估： 1.用正三角形擺法看聲音是否離箱，不能離箱的話則可說是最其本要求也不能達到，系統肯定存在很大問題，音箱時間及頻率準確度成疑，左右音箱不平衝的機會很高； 2.測試系統的音量範圍，看由最細聲到最大聲能有多寬的距離，音量太小根本聽不到音樂的變化，音量太高則超出系統能力而有clipping現象。如果可聆聽範圍太窄，則系統只能聽某類型的音樂； 3.待續 |
hkborn 正式會員116.xxx.xxx.58 |
2013-05-23 16:17 | |
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| [#42] 改善音場深度和寬度 3.三頻平衡。音樂本身是沒有分頻的，但由於喇叭單元的物理限制，聲音訊號要分拆為二分頻或更多，音箱不同單元分別重塑高中低頻，或多或少總有偏差。一般而言，三頻中低頻最難處理好，高中頻如光束，聽覺系統主要靠音量判別其存在及方向，低頻則不同，聽覺靠其相位多於音量去察覺。缺少足夠低頻，音樂整體變得薄弱，沒有質感。由於香港的狹窄環境，低頻如20Hz波長50多呎，四份一也要十多呎才夠。三頻平衡是高難度動作，所以很多人退而求其次，只玩好其中一兩段頻率，如只集中中頻的人聲，或講求氣勢包圍感的超低頻。只有由最簡單的人聲到最複雜的交嚮樂都能處理好的才算三頻均勻。 |
hkborn 正式會員 112.xxx.xxx.16 |
2013-05-23 18:29 |
| [#43] 改善音場深度和寬度 An interesting survey to see how many volts you need for your power amp to drive the speakers before it gets to the clipping point. The poll indicates that people often get too much power than what is needed for playback in domestic environment! Poll Results: I measured the test tone at: 2 volts or less 105 37.91% Between 2-5 volts 98 35.38% Between 5-10 volts 32 11.55% Between 10-20 volts 16 5.78% Over 20 volts. 26 9.39% Voters: 277. http://www.diyaudio.com/forums/multi-way/204857-test-how-much-voltage-power-do-your-speakers-need.html 最後修改時間: 2013-05-25 21:22:22 |
hkborn 正式會員 116.xxx.xxx.19 |
2013-05-25 21:21 |
| [#44] 改善音場深度和寬度 各位CHING. 小弟亦都係搅不好皇帝位9点 人声音像侧向右过 琉璃窗臺 左边是開放式廳 小弟用venture. dse4 cse. +spectral前后级十wadia cd机 請問各chinge. 可否有空光臨指導 因不懂放相 來电可wattapp大家研究64718168Mr chan THX |
omen 正式會員 203.xxx.xxx.4 |
2013-05-26 15:07 |
| [#45] 改善音場深度和寬度 各位CHING. 小弟亦都係搅不好皇帝位9点 人声音像侧向右过 琉璃窗臺 左边是開放式廳 小弟用venture. dse4 cse. +spectral前后级十wadia cd机 請問各chinge. 可否有空光臨指導 因不懂放相 來电可wattapp大家研究64718168Mr chan THX ＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝＝ 人聲音像偏右可先試試將右聲道較大看看有否改善。 或者在窗台第一反射點放cushion 以減低右邊過多反射聲。 最後修改時間: 2013-05-26 16:08:35 |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 16:07 |
| [#46] 改善音場深度和寬度 在家試試短距離近後牆的擺法，依然可以有音場的深度和寬度，就是低頻多了一些！也就是說一米的正三角型也有合理的音像！   最後修改時間: 2013-05-26 16:54:39 |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 16:44 |
| [#47] 改善音場深度和寬度 Audio Distortion By its name you know it is a measure of unwanted signals. Distortion is the name given to anything that alters a pure input signal in any way other than changing its magnitude. The most common forms of distortion are unwanted components or artifacts added to the original signal, including random and hum-related noise. A spectral analysis of the output shows these unwanted components. If a piece of gear is perfect the spectrum of the output shows only the original signal -- nothing else -- no added components, no added noise -- nothing but the original signal. The following tests are designed to measure different forms of audio distortion. THD. Total Harmonic Distortion What is tested? A form of nonlinearity that causes unwanted signals to be added to the input signal that are harmonically related to it. The spectrum of the output shows added frequency components at 2x the original signal, 3x, 4x, 5x, and so on, but no components at, say, 2.6x the original, or any fractional multiplier, only whole number multipliers. How is it measured? This technique excites the unit with a single high purity sine wave and then examines the output for evidence of any frequencies other than the one applied. Performing a spectral analysis on this signal (using a spectrum, or FFT analyzer) shows that in addition to the original input sine wave, there are components at harmonic intervals of the input frequency. Total harmonic distortion (THD) is then defined as the ratio of the rms voltage of the harmonics to that of the fundamental component. This is accomplished by using a spectrum analyzer to obtain the level of each harmonic and performing an rms summation. The level is then divided by the fundamental level, and cited as the total harmonic distortion (expressed in percent). Measuring individual harmonics with precision is difficult, tedious, and not commonly done; consequently, THD+N (see below) is the more common test. Caveat Emptor: THD+N is always going to be a larger number than just plain THD. For this reason, unscrupulous (or clever, depending on your viewpoint) manufacturers choose to spec just THD, instead of the more meaningful and easily compared THD+N. Required Conditions. Since individual harmonic amplitudes are measured, the manufacturer must state the test signal frequency, its level, and the gain conditions set on the tested unit, as well as the number of harmonics measured. Hopefully, it's obvious to the reader that the THD of a 10 kHz signal at a +20 dBu level using maximum gain, is apt to differ from the THD of a 1 kHz signal at a -10 dBV level and unity gain. And more different yet, if one manufacturer measures two harmonics while another measures five. Full disclosure specs will test harmonic distortion over the entire 20 Hz to 20 kHz audio range (this is done easily by sweeping and plotting the results), at the pro audio level of +4 dBu. For all signal processing equipment, except mic preamps, the preferred gain setting is unity. For mic pre amps, the standard practice is to use maximum gain. Too often THD is spec'd only at 1 kHz, or worst, with no mention of frequency at all, and nothing about level or gain settings, let alone harmonic count. Correct: THD (5th-order) less than 0.01%, +4 dBu, 20-20 kHz, unity gain Wrong: THD less than 0.01% THD+N. Total Harmonic Distortion + Noise What is tested? Similar to the THD test above, except instead of measuring individual harmonics this tests measures everything added to the input signal. This is a wonderful test since everything that comes out of the unit that isn't the pure test signal is measured and included -- harmonics, hum, noise, RFI, buzz ... everything. How is it measured? THD+N is the rms summation of all signal components (excluding the fundamental) over some prescribed bandwidth. Distortion analyzers make this measurement by removing the fundamental (using a deep and narrow notch filter) and measuring what's left using a bandwidth filter (typically 22 kHz, 30 kHz or 80 kHz). The remainder contains harmonics as well as random noise and other artifacts. Weighting filters are rarely used. When they are used, too often it is to hide pronounced AC mains hum artifacts. An exception is the strong argument to use the ITU-R (CCIR) 468 curve because of its proven correlation to what is heard. However, since it adds 12 dB of gain in the critical midband (the whole point) it makes THD+N measurements bigger, so marketeers prevent its widespread use. [Historical Note: Many old distortion analyzers labeled "THD" actually measured THD+N.] Required Conditions. Same as THD (frequency, level & gain settings), except instead of stating the number of harmonics measured, the residual noise bandwidth is spec'd, along with whatever weighting filter was used. The preferred value is a 20 kHz (or 22 kHz) measurement bandwidth, and "flat," i.e., no weighting filter. Conflicting views exist regarding THD+N bandwidth measurements. One argument goes: it makes no sense to measure THD at 20 kHz if your measurement bandwidth doesn't include the harmonics. Valid point, and one supported by the IEC, which says that THD should not be tested any higher than 6 kHz, if measuring five harmonics using a 30 kHz bandwidth, or 10 kHz, if only measuring the first three harmonics. Another argument states that since most people can't even hear the fundamental at 20 kHz, let alone the second harmonic, there is no need to measure anything beyond 20 kHz. Fair enough. However, the case is made that using an 80 kHz bandwidth is crucial, not because of 20 kHz harmonics, but because it reveals other artifacts that can indicate high frequency problems. All true points, but competition being what it is, standardizing on publishing THD+N figures measured flat over 22 kHz seems justified, while still using an 80 kHz bandwidth during the design, development and manufacturing stages. Correct: THD+N less than 0.01%, +4 dBu, 20-20 kHz, unity gain, 20 kHz BW Wrong: THD less than 0.01% |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 23:28 |
| [#48] 改善音場深度和寬度 IMD -- SMPTE. Intermodulation Distortion -- SMPTE Method What is tested? A more meaningful test than THD, intermodulation distortion gives a measure of distortion products not harmonically related to the pure signal. This is important since these artifacts make music sound harsh and unpleasant. Intermodulation distortion testing was first adopted in the U.S. as a practical procedure in the motion picture industry in 1939 by the Society of Motion Picture Engineers (SMPE -- no "T" [television] yet) and made into a standard in 1941. How is it measured? The test signal is a low frequency (60 Hz) and a non-harmonically related high frequency (7 kHz) tone, summed together in a 4:1 amplitude ratio. (Other frequencies and amplitude ratios are used; for example, DIN favors 250 Hz & 8 kHz.) This signal is applied to the unit, and the output signal is examined for modulation of the upper frequency by the low frequency tone. As with harmonic distortion measurement, this is done with a spectrum analyzer or a dedicated intermodulation distortion analyzer. The modulation components of the upper signal appear as sidebands spaced at multiples of the lower frequency tone. The amplitudes of the sidebands are rms summed and expressed as a percentage of the upper frequency level. [Noise has little effect on SMPTE measurements because the test uses a low pass filter that sets the measurement bandwidth, thus restricting noise components; therefore there is no need for an "IM+N" test.] Required Conditions. SMPTE specifies this test use 60 Hz and 7 kHz combined in a 12 dB ratio (4:1) and that the peak value of the signal be stated along with the results. Strictly speaking, all that needs stating is "SMPTE IM" and the peak value used. However, measuring the peak value is difficult. Alternatively, a common method is to set the low frequency tone (60 Hz) for +4 dBu and then mixing the 7 kHz tone at a value of -8 dBu (12 dB less). Correct: IMD (SMPTE) less than 0.01%, 60Hz/7kHz, 4:1, +4 dBu Wrong: IMD less than 0.01% |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 23:30 |
| [#49] 改善音場深度和寬度 IMD -- ITU-R (CCIF). Intermodulation Distortion -- ITU-R Method What is tested? This tests for non-harmonic nonlinearities, using two equal amplitude, closely spaced, high frequency tones, and looking for beat frequencies between them. Use of beat frequencies for distortion detection dates back to work first documented in Germany in 1929, but was not considered a standard until 1937, when the CCIF (International Telephonic Consultative Committee) recommend the test. [This test is often mistakenly referred to as the CCIR method (as opposed to the CCIF method). A mistake compounded by the many correct audio references to the CCIR 468 weighting filter.] Ultimately, the CCIF became the radiocommunications sector (ITU-R) of the ITU (International Telecommunications Union), therefore the test is now known as the IMD (ITU-R). How is it measured? The common test signal is a pair of equal amplitude tones spaced 1 kHz apart. Nonlinearity in the unit causes intermodulation products between the two signals. These are found by subtracting the two tones to find the first location at 1 kHz, then subtracting the second tone from twice the first tone, and then turning around and subtracting the first tone from twice the second, and so on. Usually only the first two or three components are measured, but for the oft-seen case of 19 kHz and 20 kHz, only the 1 kHz component is measured. Required Conditions. Many variations exist for this test. Therefore, the manufacturer needs to clearly spell out the two frequencies used, and their level. The ratio is understood to be 1:1. Correct: IMD (ITU-R) less than 0.01%, 19 kHz/20 kHz, 1:1, +4 dBu Wrong: IMD less than 0.01% |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 23:30 |
| [#50] 改善音場深度和寬度 S/N or SNR. Signal-To-Noise Ratio What is tested? This specification indirectly tells you how noisy a unit is. S/N is calculated by measuring a unit's output noise, with no signal present, and all controls set to a prescribed manner. This figure is used to calculate a ratio between it and a fixed output reference signal, with the result expressed in dB. How is it measured? No input signal is used, however the input is not left open, or unterminated. The usual practice is to leave the unit connected to the signal generator (with its low output impedance) set for zero volts. Alternatively, a resistor equal to the expected driving impedance is connected between the inputs. The magnitude of the output noise is measured using an rms-detecting voltmeter. Noise voltage is a function of bandwidth -- wider the bandwidth, the greater the noise. This is an inescapable physical fact. Thus, a bandwidth is selected for the measuring voltmeter. If this is not done, the noise voltage measures extremely high, but does not correlate well with what is heard. The most common bandwidth seen is 22 kHz (the extra 2 kHz allows the bandwidth-limiting filter to take affect without reducing the response at 20 kHz). This is called a "flat" measurement, since all frequencies are measured equally. Alternatively, noise filters, or weighting filters, are used when measuring noise. Most often seen is A-weighting, but a more accurate one is called the ITU-R (old CCIR) 468 filter. This filter is preferred because it shapes the measured noise in a way that relates well with what's heard. Pro audio equipment often lists an A-weighted noise spec -- not because it correlates well with our hearing -- but because it can "hide" nasty hum components that make for bad noise specs. Always wonder if a manufacturer is hiding something when you see A-weighting specs. While noise filters are entirely appropriate and even desired when measuring other types of noise, it is an abuse to use them to disguise equipment hum problems. A-weighting rolls off the low-end, thus reducing the most annoying 2nd and 3rd line harmonics by about 20 dB and 12 dB respectively. Sometimes A-weighting can "improve" a noise spec by 10 dB. The argument used to justify this is that the ear is not sensitive to low frequencies at low levels (à la Fletcher-Munson equal loudness curves), but that argument is false. Fletcher-Munson curves document equal loudness of single tones. Their curve tells us nothing of the ear's astonishing ability to sync in and lock onto repetitive tones -- like hum components -- even when these tones lie beneath the noise floor. This is what A-weighting can hide. For this reason most manufacturers shy from using it; instead they spec S/N figures "flat" or use the ITU-R 468 curve (which actually makes their numbers look worse, but correlate better with the real world). However, an exception has arisen: Digital products using A/D and D/A converters regularly spec S/N and dynamic range using A-weighting. This follows the semiconductor industry's practice of spec'ing delta-sigma data converters A-weighted. They do this because they use clever noise shaping tricks to create 24-bit converters with acceptable noise behavior. All these tricks squeeze the noise out of the audio bandwidth and push it up into the higher inaudible frequencies. The noise may be inaudible, but it is still measurable and can give misleading results unless limited. When used this way, the A-weighting filter rolls off the high frequency noise better than the flat 22 kHz filter and compares better with the listening experience. The fact that the low-end also rolls off is irrelevant in this application. (See Digital Dharma of Audio A/D Converters) Required Conditions. In order for the published figure to have any meaning, it must include the measurement bandwidth, including any weighting filters and the reference signal level. Stating that a unit has a "S/N = 90 dB" is meaningless without knowing what the signal level is, and over what bandwidth the noise was measured. For example if one product references S/N to their maximum output level of, say, +20 dBu, and another product has the same stated 90 dB S/N, but their reference level is + 4 dBu, then the second product is, in fact, 16 dB quieter. Likewise, you cannot accurately compare numbers if one unit is measured over a BW of 80 kHz and another uses 20 kHz, or if one is measured flat and the other uses A-weighting. By far however, the most common problem is not stating any conditions. Correct: S/N = 90 dB re +4 dBu, 22 kHz BW, unity gain Wrong: S/N = 90 dB |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 23:31 |
| [#51] 改善音場深度和寬度 BW. Bandwidth or Frequency Response What is tested? The unit's bandwidth or the range of frequencies it passes. All frequencies above and below a unit's Frequency Response are attenuated -- sometimes severely. How is it measured? A 1 kHz tone of high purity and precise amplitude is applied to the unit and the output measured using a dB-calibrated rms voltmeter. This value is set as the 0 dB reference point. Next, the generator is swept upward in frequency (from the 1 kHz reference point) keeping the source amplitude precisely constant, until it is reduced in level by the amount specified. This point becomes the upper frequency limit. The test generator is then swept down in frequency from 1 kHz until the lower frequency limit is found by the same means. Required Conditions. The reduction in output level is relative to 1 kHz; therefore, the 1 kHz level establishes the 0 dB point. What you need to know is how far down is the response where the manufacturer measured it. Is it 0.5 dB, 3 dB, or (among loudspeaker manufacturers) 10 dB? Note that there is no discussion of an increase, that is, no mention of the amplitude rising. If a unit's frequency response rises at any point, especially the endpoints, it indicates a fundamental instability problem and you should run from the store. Properly designed solid-state audio equipment does not ever gain in amplitude when set for flat response (tubes or valve designs using output transformers are a different story and are not dealt with here). If you have ever wondered why manufacturers state a limit of "+0 dB", that is why. The preferred condition here is at least 20 Hz to 20 kHz measured +0/-0.5 dB. Correct: Frequency Response = 20-20 kHz, +0/-0.5 dB Wrong: Frequency Response = 20-20 kHz |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 23:32 |
| [#52] 改善音場深度和寬度 Dynamic Range What is tested? First, the maximum output voltage and then the output noise floor are measured and their ratio expressed in dB. Sounds simple and it is simple, but you still have to be careful when comparing units. How is it measured? The maximum output voltage is measured as described below, and the output noise floor is measured using an rms voltmeter fitted with a bandwidth filter (with the input generator set for zero volts). A ratio is formed and the result expressed in dB. Required Conditions. Since this is the ratio of the maximum output signal to the noise floor, then the manufacturer must state what the maximum level is, otherwise, you have no way to evaluate the significance of the number. If one company says their product has a dynamic range of 120 dB and another says theirs is 126 dB, before you jump to buy the bigger number, first ask, "Relative to what?" Second, ask, "Measured over what bandwidth, and were any weighting filters used?" You cannot know which is better without knowing the required conditions. Again, beware of A-weighted specs. Use of A-weighting should only appear in dynamic range specs for digital products with data converters (see discussion under S/N). For instance, using it to spec dynamic range in an analog product may indicate the unit has hum components that might otherwise restrict the dynamic range. Correct: Dynamic Range = 120 dB re +26 dBu, 22 kHz BW Wrong: Dynamic Range = 120 dB |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 23:33 |
| [#53] 改善音場深度和寬度 Crosstalk or Channel Separation What is tested? Signals from one channel leaking into another channel. This happens between independent channels as well as between left and right stereo channels, or between all six channels of a 5.1 surround processor, for instance. How is it measured? A generator drives one channel and this channel's output value is noted; meanwhile the other channel is set for zero volts (its generator is left hooked up, but turned to zero, or alternatively the input is terminated with the expect source impedance). Under no circumstances is the measured channel left open. Whatever signal is induced into the tested channel is measured at its output with an rms voltmeter and noted. A ratio is formed by dividing the unwanted signal by the above-noted output test value, and the answer expressed in dB. Since the ratio is always less than one (crosstalk is always less than the original signal) the expression results in negative dB ratings. For example, a crosstalk spec of -60 dB is interpreted to mean the unwanted signal is 60 dB below the test signal. Required Conditions. Most crosstalk results from printed circuit board traces "talking" to each other. The mechanism is capacitive coupling between the closely spaced traces and layers. This makes it strongly frequency dependent, with a characteristic rise of 6 dB/octave, i.e., the crosstalk gets worst at a 6 dB/octave rate with increasing frequency. Therefore knowing the frequency used for testing is essential. And if it is only spec'd at 1 kHz (very common) then you can predict what it may be for higher frequencies. For instance, using the example from above of a -60 dB rating, say, at 1 kHz, then the crosstalk at 16 kHz probably degrades to -36 dB. But don't panic, the reason this usually isn't a problem is that the signal level at high frequencies is also reduced by about the same 6 dB/octave rate, so the overall S/N ratio isn't affected much. Another important point is that crosstalk is assumed level independent unless otherwise noted. This is because the parasitic capacitors formed by the traces are uniquely determined by the layout geometry, not the strength of the signal. Correct: Crosstalk = -60 dB, 20-20kHz, +4 dBu, channel-to-channel Wrong: Crosstalk = -60 dB |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 23:34 |
| [#54] 改善音場深度和寬度 a table for easy reference  |
hkborn 正式會員 203.xxx.xxx.106 |
2013-05-26 23:37 |
| [#55] 改善音場深度和寬度 多謝/hkborn兄 光試过 有d改善 左边第1 2 点都在左边喇叭發声 不能超出左侧喇叭 右边喇叭第一点已作咭哐 處哩 另左边喇叭比右边推前约4寸 |
omen 正式會員 14.xxx.xxx.27 |
2013-05-27 01:05 |
| [#56] 改善音場深度和寬度 omen hing, 皇帝位測試碟的九點，第五點在正前方，代表著左右喇叭在音量及時間準確性是否平衡。 第一點及第九點分別在左右，比較理想是在喇叭外一呎左右。如果左邊第一點不能離箱，那是右邊喇叭的弱音強度不足，反之亦然。 |
hkborn 正式會員 116.xxx.xxx.58 |
2013-05-27 13:58 |
| [#57] 改善音場深度和寬度 左边喇叭弟1/2点又離箱問題 困妖 了小弟多年 人声總係遍右 聽古典 第一小提琴遍右 后面小提琴组好像消失了 連带 木管 銅管樂器都移位 之前以為器材有問题 多年來所有器/線材 已改变 屋都变my 都未搅掂 各位CHING. 有空可否來家訪吓 我住屯門区 THX |
omen 正式會員 203.xxx.xxx.5 |
2013-05-27 14:42 |
| [#58] 改善音場深度和寬度 多谢k.c.78.及 hkborn兄提点 作晚已將对喇叭作正三再形及左边喇叭移前半呎 右边喇叭第一返射点作吸音處理 有進步 各CHING可wattapp我64718168 我發相片作為大家研究 因我不懂放相. THX |
omen 正式會員 203.xxx.xxx.15 |
2013-05-27 15:48 |
| [#59] 改善音場深度和寬度 要改善重播效果，主要有兩方面： 其一是聲學上的處理； 其二是聲音訊號在重播系統損耗減至最低。 聲學上面對的是空氣的震動，由喇叭單元表面震膜振動而產生強度不一，頻率不一的聲波而到達我們耳膜，聲音以大概每秒344米的速度移動，進入我們耳朵的，除了音箱直接的聲音外，還有四方八面反射而來的反射聲，聲音能量隨著距離而減弱，最後消失。每個聆聽空間都有這樣的殘響，我們一般用RT60來評估房間對直接聲的影響，RT60所量度的是直接聲衍生的殘響要多久才減弱至-60dB。太多會有多餘迴響，太少則令聲音死板。 聲學上要注意的是音量，頻率及相位。 在音樂重播上，我們重視的音量是指動態範圍(dynamic range)。任何聆聽環境都有或多或少的噪音，差異可以後大，深宵時可能只有人35-50dB，日間會有45－60dB。假如一般重播音量在75-90dB的範圍，我們只有相當少的動態，所以很多人覺得深夜時聽音樂特別好，就是環境噪音少而令音樂動態增加，動態愈大，對比愈大，音樂感就愈強。 聲學處理中，我們可將高、中頻看成是光束，為了減少反射聲，很多高頻單元都用上號角設計，減少上下及兩側的反射聲。通常要找出第一反射點，用一塊小鏡子放在側牆，地下及天花，在皇帝位看到單元的位置就是了。一般用吸音物直接減少其能量，或用擴散版將反射角度改變。低頻是最難處理的，主要是空間的問題。低頻波長較長，以30Hz為例，每秒為11.5米，20Hz為17米。物理上由相位而影響低頻量在房間不同點有過多或過少的現象。 |
hkborn 正式會員 116.xxx.xxx.58 |
2013-05-27 21:28 |
| [#60] 改善音場深度和寬度 photo 1  |
l5a 正式會員 14.xxx.xxx.55 |
2013-05-29 01:12 |
