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音频参数测试的方法以及步骤

音频参数测试的方法以及步骤

测试中发现的问题及对策
• 关于STMR
• 从送话器到受话器的声传输称之为侧音。
• 使用META工具修改Side Tone的数值或者进入工程 模式在音频中的正常中找到Side Tone选项修改其数值使其Βιβλιοθήκη 在13 +/- 5 dB 内。
测试中发现的问题及对策
• 针对:稳定度储备
对手持机,将手持机防在坚硬平面上,传感器面向平面。如有音 量控制器,将其置为最大。 注要控制手机的接收和发送的增益,以使的手机不要啸叫。
式在音频中的正常中找到Speech选项修改其数值使
其在2 +/- 3 dB的范围内。
测试中发现的问题及对策
• 针对SFR 发送频响
MIC的质量
主要受MIC本身和手机的物理结构影响。另外于电路也有一定的关系。
具体的调试方法在META中将给出。
测试中发现的问题及对策
• 针对RFR 接收频响
主要受本身和手机的物理结构影响。另外于电路也有一定的关系。
不同仪表之间的差异
不同仪表之间的差异
GSM 11.10 30. 1 R ec eiv ing f requency res pons e AVGL//dRB[FPa/V] 30
20
10
0
-10
R CV(2)-R ef
lower
-20
upper
-30 120 200 300 400 f / Hz 1000 1600 2400 4000
dB relative to ARL
35 dB -30 dB -20 dB -10 dB
0 dB 7 dB 10 dB
Level ratio
17,5 dB 22,5 dB 30,7 dB 33,3 dB 33,7 dB 31,7 dB 25,5 dB

音频客观测量指标概念(全)

音频客观测量指标概念(全)

音频客观测量指标概念音频指标简介及测试原理方法音频指标测试均是针对有输入和输出的设备而言,就是声音信号经过了一个通道以后,输出与输入之间的差别。

两者差别越小那么性能越好,而且在一般情况下声音经过某一个通道或某一系统后,一般都有对原信号的放大和衰减。

信噪比、失真率、频率响应这三个指标是音响器材的“基础指标”或“基本特性”,我们在评价一件音响器材或者一个系统水准之前,必须先要考核这三项指标,这三项指标中的任何一项不合格,都说明该器材或者系统存在着比较重大的缺陷1、信噪比SNR(Signal to Noise Ratio):(1)简单定义:狭义来讲是指放大器的输出信号的电压与同时输出的噪声电压的比,常常用分贝数表示,设备的信噪比越高表明它产生的杂音越少。

一般来说,信噪比越大,说明混在信号里的噪声越小,声音回放的音质量越高,否则相反。

信噪比一般不应该低于70dB,高保真音箱的信噪比应达到110dB以上。

音频信噪比是指音响设备播放时,正常声音信号强度与噪声信号强度的比值(2)计算方法:信噪比的计量单位是dB,其计算方法是10LG(PS/PN),其中Ps和Pn 分别代表信号和噪声的有效功率,也可以换算成电压幅值的比率关系:20LG(VS/VN),Vs和Vn分别代表信号和噪声电压的“有效值”。

(3)测量方法:信噪比通常不是直接进行测量的,而是通过测量噪声信号的幅度换算出来的,通常的方法是:给放大器一个标准信号,通常是0.775Vrms或2Vp-p@1kHz,调整放大器的放大倍数使其达到最大不失真输出功率或幅度(失真的范围由厂家决定,通常是10%,也有1%),记下此时放大器的输出幅Vs,然后撤除输入信号,测量此时出现在输出端的噪声电压,记为Vn,再根据SNR=20LG(Vn/Vs)就可以计算出信噪比了. 或者是10LG(PS/PN),其中Ps和Pn分别代表信号和噪声的有效功率计权:这样的测量方式完全可以体现设备的性能了。

《数字音频处理技术》在线理论测试

《数字音频处理技术》在线理论测试

《数字音频处理技术》在线理论测试适用于数字媒体技术2021级一.单选题(共20题,每题0.5分)1.人耳所能听到的声音波长范围是多少?() [单选题] *A.0.017m~17m(正确答案)B.0.016m~16mC.0.015m~15mD.0.014m~14m2.要录制来自话筒的声音,需要将话筒的连接线与声卡的()接口连接? [单选题] *A.Line inB.Mic(正确答案)C.SpkD.Midi3.将模拟音频转换为数字音频,又将数字音频转换为模拟音频的设备是()。

[单选题] *A.音响B.音箱C.声卡(正确答案)D.PCI卡4.在常温下,声音在空气中的传播速度是?() [单选题] *A.120m/sB.180m/sC.260m/sD.340m/s(正确答案)5.信号源S={S1,S2,S3,S4,S5,S6,S7},对应的概率为P={0.2,0.19,0.18,0.17,0.15,0.1,0.01},对其进行霍夫曼编码后的平均码字长度为()[单选题] *A.3.38B.2.72(正确答案)C.1.96D.3.226.实现侧链效果需要对音频轨道添加什么效果器?() [单选题] *A.调制效果B.动态处理(正确答案)C.多普勒换档器D.母带处理7.下面关于Audition的描述错误的是() [单选题] *A.在多轨编辑中可以实现与视频的集成B.在多轨编辑中可以实现多条音频轨道的混缩C.可在单轨编辑下实现侧链效果(正确答案)D.可在单轨编辑下查看波形的频谱特性8.对立体声的某波形声音进行数字化时,采样频率为4KHz,量化位数为16位,则其未压缩时的数据传输速率为()? [单选题] *A.16kbpsB.32kbpsC.64kbpsD.128kbps(正确答案)9.CD音频的标准采样率是多少()? [单选题] *A.96KHzB.22.05KHzC.48KHzD.44.1KHz(正确答案)10.录制来自外接设备的声音时,录制声音来源应该选择()设备。

音频测试指标与测试经验

音频测试指标与测试经验

FTA音频测试及测试经验厦门厦新移动通讯有限公司研发中心测试部厦门海沧新阳工业区厦新电子城 361022狄德海didehai@Tel: 86-0592-*******-32741. 音频测试项目在FTA音频测试中音频测试的项目有30.1,30.2,30.3,30.4,30.5.1,30.6.2,30.7.1参考GSM11.10注意事项所有的测试项目应在同一天的测试时间里通过但每一项的测试可以有多次测试直到测试通过为止30.1发送频率响应Sending Frequency Response30.1.1 定义发送灵敏度/频率响应用DB表示是指输入测试单音频时数字音频接口DAI的输出电平以PCM比特流代表与仿真嘴中的输入声压之比30.1.2 指标发送灵敏度/频率响应MRP-ÆDAI应处于表1给出的框罩内在对数频率/线形DB灵敏度坐标上对表1中的间断点之间画直线得到一个框罩如图1模板如下表1 发送灵敏度/频率响应Frequency (Hz) Upper Limit (dB) Lower Limit (dB)100 -12200 0300 0 -121000 0 -62000 4 -63000 4 -63400 4 -94000 030.1.3 测试方法a) 将手机装在LRGP中耳承密合于仿真耳的刃形边缘上b) 用仿真嘴在嘴参考点MRP送一个声压为 – 47dBPa的纯单音c) MS的DAI连接SS操作模式为音频设备及A/D D/A的测试d) 在100Hz~4000Hz频段内用1/2倍频间隔进行测试e) 在各个频率测DAI处PCM比特流代表的输出电平30.2 发送响度评定值Sending Loudness Rating SLR30.2.1 定义SLR是一种基于客观单音测试的表示发送频率响应的方法30.2.2 指标8 3 DB经验低dB值对应大的响度5dB对应最大的响度11dB代表最小的响度测试时通过调整手机的麦克风到人工嘴的距离使测试的值达到标准如果比标准值大则需调小手机麦克风到人工嘴的距离若比标准值小则调大其距离30.3 接收频率响应Receiving Frequency Response30.3.1 定义接收灵敏度/频率响应用DB表示是指仿真耳中的输出声压与DAI处PCM比特流代表的输入电平之比30.3.2 指标接收灵敏度/频率响应DAI至ERP应处于表2给出的框罩内在对数频率/线形DB灵敏度坐标上对下表中的间断点之间画直线得出框罩*的极限处于间断点之间所画的直线上30.3.3 测试方法a) 将手机装在LRGP中耳承应密合于仿真耳的刃行边缘上b) MS的DAI连接SS工作模式为音响设备与A/D D/A的测试c) SS通过DAI给MS发送一个相当于-16 dBm0纯单音的PCM比特流d) 在100HZ~40000HZ频段以1/2倍频间隔进行测试e) 在各个频率测仿真耳中耳参考点—ERP的声压经验手机与人工耳的密封性要整好表2 接收灵敏度/频率响应Frequency (Hz) Upper Limit (dB) Lower Limit (dB)100 -12200 0300 2 -7500 * -51000 0 -53000 2 -53400 2 -104000 230.4接收频率响应Receiving Loudness Rating RLR30.4.1 定义RLR是一种基于客观单音测试的表示接收频率响应的方法30.4.2 指标对于接收音量控制器对至少某一控制值RLR应满足dB当控制器置为最大时应不小于dB经验dB值较小对应较大的音量值dB代表最大发音量dB代表最小发音量在测试中通过调整手机的通话音量使达到标准值如果还不行就对SPEAK的发音孔进行大小调整比如用橡皮泥堵住其中的一个孔等等30.5.1 侧音掩蔽评定值(Side one Masking Rating, STMR30.5.1.1 定义侧音掩蔽评定值是基于客观单音的测试表示仿真嘴至仿真耳的通路损耗30.5.1.2 指标135dB经验如果STMR测试值与标准值相差较大则需要通过软件改变其参数若相差不大则可以通过调整音量来解决譬如比标准值大则需增大通话音量若比标准值小则需要调小通话音量30.6.2 稳定度储备Stability Margin30.6.2.1 定义稳定度储备是指产生震荡时需要的基准话音编译器的来去通路间插入的增益也就是用来反映手机音频是否容易出现自激振荡30.6.2.2 指标最小稳定度储备应为6dB并检测不到音频震荡30.6.2.3 测试方法a) 在中的基准话音编译器的来去通路的环路中插入一个相当于最小稳定度边际的增益并启动任一声回波控制器b) 将一个符合原建议.的测试信号在基准话音编译器的数字输入端插入环路观察稳定度测试信号的电平为dBm0,持续时间为c) 若存在用户控制的音量控制器应设置为最大值d) 将手机放在坚硬的平面上传感器面向平面经验稳定度储备一般情况下都会通过30.7.1 发送失真Sending Distortion30.7.1.1 定义发射信号与总失真之比是对发射设备不包括话音编译器线形度的量度30.7.1.2 指标用噪声加权滤波器在处测得的信号与总失真功率之比应高于表给出的极值30.7.1.3 测试方法a) 将手机装在中耳承要密合在仿真耳的刃行边缘上b) 的连接工作模式为音响设备及的测试c) 在中输入一个正弦波信号频率介于之间调节此信号的电平直到处输出的比特流等效为dBm0此时处的信号电平及为声参考电平d) 输入测试信号其电平相对于分别为-35dB -30dB -25dB -20dB -15dB -10dB -5dB 0dB 5dB 10dBe) 在每一个信号电平上用噪声加权滤波器测处信号于总失真的功率之比测试过程中声压不得超过dBPa表测出信号与总失真功率之比dB relative to ARL Level ratio-35 dB 17,5 dB-30 dB 22,5 dB-20 dB 30,7 dB-10 dB 33,3 dB0 dB 33,7 dB7 dB 31,7 dB10 dB 25,5 dB经验由于发送失真测试具有随机性只要在测试频点上的测试值与标准值相差不超过个dB多测试几次就会通过。

音频测试参数

音频测试参数

Audio Specifications-Audio Specifications• Audio Distortion• THD - Total Harmonic Distortion• THD+N - Total Harmonic Distortion + Noise • IMD – SMPTE - Intermodulation Distortion • IMD – ITU-R (CCIF) - Intermodulation Distortion • S/N or SNR - Signal-To-Noise Ratio • EIN - Equivalent Input Noise• BW - Bandwidth or Frequency Response • CMR or CMRR - Common-Mode Rejection • Dynamic Range• Crosstalk or Channel Separation • Input & Output Impedance • Maximum Input Level • Maximum Output Level • Maximum Gain • Caveat EmptorDennis BohnRane CorporationRaneNote 145© 2000 Rane CorporationIntroductionObjectively comparing pro audio signal processing products is often impossible. Missing on too many data sheets are the conditions used to obtain the published data. Audio specifica-tions come with conditions. Tests are not performed in a vacuum with random parameters. They are conducted using rigorous procedures and the conditions must be stated along with the test results.To understand the conditions, you must first understand the tests. This note introduces the classic audio tests used to charac-terize audio performance. It describes each test and the condi-tions necessary to conduct the test.Apologies are made for the many abbreviations, terms and jargon necessary to tell the story. Please make liberal use ofRane’s Pro Audio Reference (/digi-dic.html ) to help decipher things. Also, note that when the term impedance is used, it is assumed a constant pure resistance, unless otherwise stated.The accompanying table (back page) summarizes common audio specifications and their required conditions. Each test is described next in the order of appearance in the table.Audio DistortionBy 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 DistortionWhat is tested? A form of nonlinearity that causes unwant-ed 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 multi-plier, 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. Perform-ing 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 fre-quency. Total harmonic distortion (THD) is then defined as the ratio of the rms voltage of the harmonics to that of the funda-mental 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 dif-ficult, 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 mean-ingful and easily compared THD+N.Required Conditions. Since individual harmonic ampli-tudes 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 overthe 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 gainWrong:THD less than 0.01%THD+N. Total Harmonic Distortion + NoiseWhat 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 ofall signal components (excluding the fundamental) over some prescribed bandwidth. Distortion analyzers make this measure-ment 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 harmon-ics 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 mea-surements. 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. An-other argument states that since most people can’t even hear the fundamental at 20 kHz, let alone the second harmonic, thereis 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 BWWrong:THD less than 0.01%IMD – SMPTE. Intermodulation Distortion – SMPTE MethodWhat is tested? A more meaningful test than THD, inter-modulation 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 intermodulationAudio Specifications-distortion analyzer. The modulation components of the upper signal appear as sidebands spaced at multiples of the lower fre-quency 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 dBuWrong:IMD less than 0.01%IMD – ITU-R (CCIF). Intermodulation Distortion – ITU-R MethodWhat 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 fre-quencies for distortion detection dates back to work first docu-mented in Germany in 1929, but was not considered a standard until 1937, when the CCIF (International Telephonic Consulta-tive Committee) recommend the test. [This test is often mistakenly referred to as the CCI R method (as opposed to the CCI F method).A mistake compounded by the many correct audio references to the CCI R 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 dBuWrong: IMD less than 0.01%S/N or SNR. Signal-To-Noise RatioWhat 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 isto 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 us-ing 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” measure-ment, 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 appropri-ate 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 respective-ly. Sometimes A-weighting can “improve” a noise spec by 10 dB.The argument used to justify this is that the ear is not sensi-tive to low frequencies at low levels (´ 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 con-verters 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 the RaneNote 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.Audio Specifications-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 how-ever, the most common problem is not stating any conditions.Correct: S/N = 90 dB re +4 dBu, 22 kHz BW, unity gainWrong: S/N = 90 dBEIN. Equivalent Input Noise or Input Referred NoiseWhat is tested? Equivalent input noise, or input referred noise, is how noise is spec’d on mixing consoles, standalone mic preamps and other signal processing units with mic inputs. The problem in measuring mixing consoles (and all mic preamps)is knowing ahead of time how much gain is going to be used. The mic stage itself is the dominant noise generator; therefore, the output noise is almost totally determined by the amount of gain: turn the gain up, and the output noise goes up accordingly. Thus, the EIN is the amount of noise added to the input signal. Both are then amplified to obtain the final output signal.For example, say your mixer has an EIN of –130 dBu. This means the noise is 130 dB below a reference point of 0.775 volts (0 dBu). If your microphone puts out, say, -50 dBu under normal conditions, then the S/N at the input to the mic preamp is 80 dB (i.e., the added noise is 80 dB below the input signal). This is uniquely determined by the magnitude of the input signal and the EIN. From here on out, turning up the gain increases both the signal and the noise by the same amount.How is it measured? With the gain set for maximum and the input terminated with the expected source impedance, the output noise is measured with an rms voltmeter fitted with a bandwidth or weighting filter.Required Conditions. This is a spec where test conditions are critical. It is very easy to deceive without them. Since high-gain mic stages greatly amplify source noise, the terminating input resistance must be stated. Two equally quiet inputs will measure vastly different if not using the identical input imped-ance. The standard source impedance is 150 Ω. As unintuitive as it may be, a plain resistor, hooked up to nothing, generates noise, and the larger the resistor value the greater the noise. It is called thermal noise or Johnson noise (after its discoverer J. B. Johnson, in 1928) and results from the motion of electron charge of the atoms making up the resistor. All that moving about is called thermal agitation (caused by heat – the hotter the resistor, the noisier).The input terminating resistor defines the lower limit of noise performance. In use, a mic stage cannot be quieter than the source.A trick which unscrupulous manufacturers may use is to spec their mic stage with the input shorted – a big no-no, since it does not represent the real performance of the preamp.The next biggie in spec’ing the EIN of mic stages is band-width. This same thermal noise limit of the input terminating resistance is a strong function of measurement bandwidth. For example, the noise voltage generated by the standard 150 Ω input resistor, measured over a bandwidth of 20 kHz (and room temperature) is –131 dBu, i.e., you cannot have an operating mic stage, with a 150 Ω source, quieter than –131 dBu. However, if you use only a 10 kHz bandwidth, then the noise drops to –134 dBu, a big 3 dB improvement. (For those paying close attention: it is not 6 dB like you might expect since the bandwidth is half. It is a square root function, so it is reduced by the square root of one-half, or 0.707, which is 3 dB less).Since the measured output noise is such a strong functionof bandwidth and gain, it is recommended to use no weighting filters. They only complicate comparison among manufacturers. Remember: if a manufacturer’s reported EIN seems too good to be true, look for the details. They may not be lying, only using favorable conditions to deceive.Correct: EIN = -130 dBu, 22 kHz BW, max gain, Rs = 150 ΩWrong: EIN = -130 dBuBW. 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 frequen-cy (from the 1 kHz reference point) keeping the source ampli-tude 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) maybe even 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 funda-mental 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 dBWrong: Frequency Response = 20-20 kHzCMR or CMRR. Common-Mode Rejection or Common-Mode Rejection RatioWhat is tested? This gives a measure of a balanced input stage’s ability to reject common-mode signals. Common-mode is the name given to signals applied simultaneously to both inputs. Normal differential signals arrive as a pair of equal voltages that are opposite in polarity: one applied to the positive input and the other to the negative input. A common-mode signal drives both inputs with the same polarity. It is the job of a well designed bal-Audio Specifications-anced input stage to amplify differential signals, while simulta-neously rejecting common-mode signals. Most common-mode signals result from RFI (radio frequency interference) and EMI (electromagnetic interference, e.g., hum and buzz) signals induc-ing themselves into the connecting cable. Since most cables con-sist of a tightly twisted pair, the interfering signals are induced equally into each wire. The other big contributors to common-mode signals are power supply and ground related problems between the source and the balanced input stage.How is it measured? Either the unit is adjusted for unity gain, or its gain is first determined and noted. Next, a generator is hooked up to drive both inputs simultaneously through two equal and carefully matched source resistors valued at one-half the expected source resistance, i.e., each input is driven from one-half the normal source impedance. The output of the bal-anced stage is measured using an rms voltmeter and noted. A ratio is calculated by dividing the generator input voltage by the measured output voltage. This ratio is then multiplied by the gain of the unit, and the answer expressed in dB.Required Conditions. The results may be frequency-depen-dent, therefore, the manufacturer must state the frequency tested along with the CMR figure. Most manufacturers spec this at 1 kHz for comparison reasons. The results are assumed constant for all input levels, unless stated otherwise.Correct: CMRR = 40 dB @ 1 kHzWrong: CMRR = 40 dBDynamic RangeWhat 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 care-ful when comparing units.How is it measured? The maximum output voltage is mea-sured 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 maxi-mum 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, us-ing 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 Crosstalk or Channel SeparationWhat 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 chan-nel 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-channelWrong: Crosstalk = -60 dBInput & Output ImpedanceWhat is tested? Input impedance measures the load that the unit represents to the driving source, while output impedance measures the source impedance that drives the next unit.How is it measured? Rarely are these values actually mea-sured. Usually they are determined by inspection and analysis of the final schematic and stated as a pure resistance in Ωs. Input and output reactive elements are usually small enough to be ignored. (Phono input stages and other inputs designed for specific load reactance are exceptions.)Required Conditions. The only required information is whether the stated impedance is balanced or unbalanced (bal-anced impedances usually are exactly twice unbalanced ones). For clarity when spec’ing balanced circuits, it is preferred to state whether the resistance is “floating” (exists between the two lines) or is ground referenced (exists from each line to ground).The impedances are assumed constant for all frequencies within the unit’s bandwidth and for all signal levels, unlessAudio Specifications-。

音频测试方案

音频测试方案
第2篇
音频测试方案
一、前言
为确保音频设备在交付用户使用前具备卓越的性能和稳定的品质,本方案针对音频设备的功能、性能、可靠性和用户体验等方面进行系统性的测试。本方案旨在规范测试流程、方法和标准,为产品质量提升提供科学依据。
二、测试目标
1.确保音频设备的功能齐全,操作便捷;
2.评估音频设备的性能指标,满足国家和行业标准;
4.用户体验测试:模拟用户实际使用场景,评估音频设备的易用性、舒适度等。
六、测试流程
1.准备阶段:收集测试设备、标准、工具等;
2.测试计划:制定详细的测试计划,明确测试项目、方法、时间等;
3.测试执行:按照测试计划进行测试,记录测试数据;
4.数据分析:对测试数据进行分析,发现问题并提出改进措施;
5.撰写测试报告:整理测试结果,撰写测试报告;
3.用户需求:根据用户实际使用场景和需求进行测试。
五、测试方法
1.功能测试:检查音频设备的各项功能是否正常,如播放、暂停、停止、上一曲、下一曲等;
2.性能测试:评估音频设备的性能指标,如频率响应、失真度、信噪比等;
3.稳定性和可靠性测试:对音频设备进行长时间连续工作测试,检查设备在不同环境条件下的稳定性;
(3)信噪比:测试音频设备的信噪比,评估其抗干扰能力。
3.稳定性和可靠性测试:
(1)长时间连续工作测试:检查音频设备在长时间连续工作下的性能稳定性;
(2)环境适应性测试:检查音频设备在不同温度、湿度等环境条件下的性能稳定性。
4.用户体验测试:
(1)易用性测试:评估音频设备的操作便捷性、界面友好性等;
十、总结
本音频测试方案从功能、性能、可靠性和用户体验等方面对音频设备进行全面评估,旨在确保产品在交付用户前具备卓越的品质。通过严谨的测试流程和方法,为产品质量提升提供科学依据,助力企业提高市场竞争力。

音频测试参数详解

音频测试参数详解

一、SLR=Lg(标准信号/麦克风接收到的信号);当测试结果大于11dB时,适当增加麦克风电路增益;当测试结果小于5dB时,适当降低麦克风电路增益;二、RLR=Lg(标准信号/听筒发出的音频信号)当测试结果小于-1dB时,适当降低听筒电路增益;当测试结果大于5dB时,适当增加听筒电路增益;三、SFR麦克风的质量,质量的好坏直接影响SFR的测试结果;手机物理结构;基带电路;四、RFR1>听筒的质量直接反映在测试结果上;2>听筒的声学中心如果与其物理中心不一致,也会影响测试结果;3>不正确的测试方法会导致测试结果的不可比;4>RF模式和DAI模式的不同,对测试结果有一定的影响;五、STMR=Lg(仿真嘴发出的音频信号/听筒发出的仿真嘴发出的音频信号)1>从麦克风到听筒的声传输称为侧音(Side tone);2>电话的侧音通道就是发话者讲话时能听到自己声音的一种通道,其他侧音通道还有头传导通道和嘴与耳朵之间经过耳承泄漏形成的声通道。

这些附加侧音通道的存在影响了用户对侧音的感觉,因此也影响了他对侧音的反映。

3>侧音从几个方面影响电话传输质量。

如果侧音损耗太小,则回到自己耳朵的话音声级太响;另一方面,若侧音损耗太大,还会使发话者趋于降低其讲话的声级或形成对方误以为发话者的麦克风远离嘴巴,从而使收话者的受听声级下降。

六、失真1>当系统的输入与输出不呈线性关系时,就要产生非线性失真;2>非线性失真对数据传输而言比语音传输更重要,但是对语音传送也很重要;3>量化失真:在数字系统中,当模拟信号被抽样,再把每个抽样信号编码为有限数字时就会出现量化失真。

把原始信号与量化后又复原的信号作比较,将差异叫做量化失真和非线性失真。

现在采用编码公式A律或者U律PCM都采用接近对数的压扩率。

七、稳定度余量将手机放在坚硬平面上,传感器面向平面,如果有音量控制器,将其置为最大。

音频性能测试指引

音频性能测试指引

音频性能测试用例一、仪器设备:VA-2230音频分析仪;负载(4欧或8欧);32欧耳机负载二、准备工作:2.1、对即将测试的机器升级最新软件,并确认喇叭和耳机均可以正常输出。

2.2、将测试用音频文件拷贝到机器中,2.3、接线:左声道的两个红线分别接喇叭(或耳机)的左声道输出,其余两根黑线接主板上的地。

右声道的两个红线分别接喇叭(或耳机)的右声道输出的,其余两根黑线接主板上的地。

以上测试需保证喇叭和耳机均已连接标准的负载。

三、初始设置:3.1、打开 VA-2230 音频分析仪,待仪器预热 15 分钟后进行以下测试3.2、按 VA-2230 音频分析仪的←↑按钮或→↓按钮,选中 Input 将输入耦合阻抗设定为:10KΩ, 耦合方式设定为: balance(即平衡模式)如下图:注意:数字功放选择balance(即平衡模式),模拟功放选择unbalance(即非平衡模式)。

3.3、按 VA-2230 音频分析仪的←↑按钮或→↓按钮, 选中 SP,并将其设定为 Slow, 将 SS设定为 1.5s;四、各测试项测试方法及步骤:3.1、最大输出功率A、按 VA-2230 音频分析仪的←↑按钮或→↓按钮,将 HPF,PSO 设置为 OFF,LPF设置为20KHz(模拟功放LPF要设置为OFF)。

B、播放机器中的《08-1KHz-0dB》音频文件,并将音量调到最大。

按音频分析仪(中部上端)的AC-V按钮,音频分析仪屏幕左上方若出现ACV,表明已经选中,调节按钮选中UNIT 项,按钮F3 切换为V。

此时屏幕上显示的为左右声道输出的有效值。

最大输出功率必须满足总谐波失真的指标,如果总谐波失真超标,需将音量调小重新确认最大输出幅值。

总谐波失真测试方法见3.4。

注:屏幕左上方会显示Freq=1000Hz,或者频率很接近1000Hz。

如果此处未显示出数字,说明设置有误。

C、输出功率=输出幅值 /负载阻抗。

D、标准:不要超过喇叭或耳机的额定功率3.2、频率响应A、按照3.1 调节好仪器,播放机器中《08-1KHz-0dB》音频文件。

参数测试方法

参数测试方法

目录Content3. 电流要求 Current Requirement静态电流 Quiescent Current最小电流和最大电流 Minimum Current and Maximum Current4. 音频模式输出功率 Output Power、左右平衡控制 Effectivity Of Balance Control、前后平衡控制 Effectivity Of Fader Control、低音控制 Bass Control、高音控制 Treble Control、响度控制 Effectivity Of Loudness、剩余噪音 Residual Noise On Loudspeaker、源分隔度 Source Separation、喇叭增益 GAIN ON .、电平匹配 Output Level Differences5 AM范围内的特殊要求Special requirements in the AM rangeAM一般要求AM General RequirementsAM 频率范围:AM RangesAM中频频率fi, AM Intermediate frequency fiAM信噪比 AM Signal to noise ratioAM噪限灵敏度(有用灵敏度)(E'N) AM Noise limited sensitivityAM增益控制灵敏度(Merit Sensitivity)AM锁台灵敏度 AM Search tuning sensitivityAM选择性 AM SelectivityAM中频抑制 AM IF-rejection ratioAM镜频抑制 AM Image rejection ratioAM假响应抑制 AM Spurious response rejection ratioAM全音频响应 AM Overall AF responseAM大信号支持能力 AM Large signal handling capabilityAM射频失真 AM RF distortion6 FM范围内的特殊要求 Special requirements in the FM rangeFM一般要求 FM General RequirementsFM 频率范围:FM中频频率fiFM噪限灵敏度(有用灵敏度)(E'N)Noise-limited sensitivity (E'N).1 FM信噪比Signal to noise ratioFM台间噪音 Interstation noiseFM-3dB限幅点 FM-3dB Limiting pointFM锁台灵敏度 FM Search tuning sensitivityFM锁台时间 FM Tuning timeFM选择性FM SelectivityFM中频抑制 FM IF-rejection ratioFM镜频抑制 FM Image rejection ratioFM全音频响应 FM Overall AF responseFM大信号支持能力 FM Large signal handling capabilityFM射频失真 FM RF distortionFM声道分离度 FM Channel separationFM10dB声道分离度(SDS) FM signal dependent stereoFM导频和副载波抑制 Suppression7、立体声要求 Stereo requirement声道平衡 Channel balance单声道-立体声转换点 Mono-Stereo transition导频和副载波抑制Pilot frequency and subcarrier suppression 7. 磁带放音模式 Tape Mode带速误差 Tape Speed Deviation速度控制效益 Speed Control Effectivity抖晃 Wow and Flutter频率响应 Frequency Response输出电平差异 Output Level Difference立体声声道分隔度 Channel Separation串音 Crosstalk失真 Distortion信噪比 S/N Ratio8. CD放音模式频率响应Frequency Response立体声串音 Stereo Crosstalk总谐波失真 Total Harmonic Distortion信噪比 S/N Ratio噪音电平搜索 Noise Lever During Search去加重 De-emphasis备注:参数测试部分所有的测试若在具体条目下无特殊要求,都要求在室温23±5℃,测试电压14±,4×4Ω负载条件下测量,E'为试验机直接接到的信号的电平。

音质测试方法讲解

音质测试方法讲解

放音质量测试目的:测试耳机、扬声器的相关指标输出参数是否符合标准测试条件:1、所有指标均在稳态(背光关闭、无操作20秒后)下测量, 测试文件为48KS/s_16bit_PCM_wav格式(从测试工具中生成,点击所需要测试的指标,然后保存即可)2、声压级测试条件:●消音室环境●扬声器与声压仪水平距离30cm,垂直距离0cm●用扬声器以最大音量播放2kHz正弦波测试方法:●消音室环境●扬声器与声压仪水平距离30cm,垂直距离0cm●用扬声器以最大音量播放2kHz正弦波期望结果:扬声器输出其它测试项:对扬声器的稳定性测试:用cool edit制作一段白噪、粉噪声音文件放在设备中播放12小时、24小时,再次验证扬声器振膜是否有衰老,对比第一次的测试结果声音效果测试:用不同的音乐分别测试扬声器的高、中、低频段最大音量播放时的声音效果(用试音音频文件:)测试工具介绍:RightMark Audio Analyzer 6.2 (此软件是一款声卡等音频设备的评测软件,功能相当全面,比如:频率响应、噪声水平、动态范围、总谐波失真、立体声分离度以及互调失真等指标它均可测定。

测定出的数据以曲线图表显示,非常直观,支持生成网页形式的评测结果,便于对比查看和保存。

)关键词解释:1.1Frequency Response频率响应(FR:Frequency Response):频率响应是指将一个以电压输出的音频信号,产生的声压随频率的变化而发生增大或衰减、相位随频率而发生变化的现象,这种声压和相位与频率的相关联的变化关系称为频率响应。

简单理解,频率响应,反应播放设备的各个频率的声音信号的信号相对大小是否保持面貌理想情况下,频响曲线应是一条直线。

好的频响曲线在每隔一个频率点都能输出稳定足够的信号,不同频率点彼此之间的信号大小均一样,然而在低频与高频部分信号的重建比较困难,所以这两个频段都会出现衰减现象。

输出品质越好的频响曲线就越平直,反之不但在高频和低频处衰减的很快,一般频段也可能出现抖动现象。

广播中心技术指标测量

广播中心技术指标测量
接;
④除了特殊的规定外,被测试的音频信号通道中的均衡器、 滤波器等均应设置在其幅频特性为平直的位置;
⑤调音台设置在额定工作电平的状态。
3.参考电平:
国家标准《广播声频通路技术指标测量方法》 (GBT15943-1995)中规定,参考电平是“使声频设 备的音量指示达到0VU100%)刻度时的电平”。在广 播中心播控系统中,采用0VU对应+4dBu作为播出系统 的参考电平。参考电平也称为校准电平,校准电平使 用的标准测量信号是频率为1KHz、电压有效值为 1.228V的单频正弦波信号。
技术指标测量的重要性.
对广播中心系统的核心设备和节目播出通道进行测试, 对于新建系统是衡量系统设计和施工质量的重要依据; 对于广播中心系统的技术维护工作,是保持系统经常 处于良好运行状态的重要手段。
对广播中心系统来说,必须定期对在线播出设备和系 统、对播出环节信号电平进行技术测试,以保证系统 工作状态正常。
GITITAL I/O
测试中信号发生器指标的选择
(1)、模拟信号发生器 线路输入电平:+4dBu, 话放输入电平: -55 ~ -50dBu, 参考频率:1000Hz, 发生器输出阻抗:40 欧姆 BAL。
(2)、数字信号发生器 嵌入音频电平:-20dBFS, 参考频率:997Hz,24bit,没有抖晃。 模进数出:参考频率997Hz
广播中心技术指标测试中的有关术语
1.参考频率:广播中心系统在进行系统和设备的技术性能 和技术参数的测试时,作为参考点的频率,模拟系统测试 以1000Hz为参考频率;数字系统测试以997Hz为参考频率。
2.正常工作状态:广播中心的系统和设备在满足下列条件 时,即为正常工作状态。
①符合规定的工作条件; ②被测试的系统或设备的输出端接规定的负载阻抗; ③被测试的系统或设备的不同的端口按厂方的规定进行连

模拟音频参数和测试.doc

模拟音频参数和测试.doc

模拟音频参数和测试1. 基本单位和概念dBu 以0.775V (有效值)为基准电压时的电压电平单位。

表示为:dBu=20lg(v/0.775)dBu 的计算只考虑电压电平本身,而不考虑与相应的电功率电平之间的关系,不考虑阻抗是否为600Ω。

---参照GY/T 192-2003dBu 采用接近0的源阻抗和接近无穷大的负载阻抗!基准信号的电平为0.775V RMS dBu=20log(Vx/0.775),Vx=0.775*10(Dbu/20),0.775V RMS 对应的电阻为600ohm,即1mW 在600ohm 产生0.77459的电压220.7750.001600U P WR ==注意dBu 表示的是电压值,在音频上并不是以1uV 作为基准电压,而是以0.775V RMS 作为基准 所以dBu 换算出来是RMS 值Vrms有效值,均方根值,正弦波时,均方根值Vrms为振幅Vm的0.707倍,为峰峰值的0.707/2倍Vpp峰峰值Vpp=2*Vm=2.828VrmsdBFs(dB below digital Full Scale)以满刻度的量值为0dB,常用于各种特性曲线上;数字音频信号测量中经常使用到单位“dbFS”。

0dbFS既是指满刻度的数字音频参考电平,即“数字满刻度电平”,它是指在数字域的音频系统中,A/D或D/A转换器可能达到的“数字过载”之前的最大可编码模拟信号电平。

0dbFS为数字音频信号最高峰的绝对值,与16bit线性编码PCM信号对应的最高值为7FFF(16进制),最高负值电平为8000(16进制),十进制数为32767。

不同国家对数字设备满度电平值OdBFS所对应的电平模拟信号的电平值不尽相同,目前还没有这个标准数字码的国际标准,常见的是SMPTE。

(美国电影电视工程师学会)和EBU(欧洲广播联盟)推荐的两个方案。

SMPTE推荐的转换基准规定为对于16bit的PCM声音信号,频率为lkHz的模拟正弦波信号的正、负峰值使A/D转换器分别产生OCCD,F333数字码时的幅度为参考电平。

音频测试参数解析

音频测试参数解析

Frequency Response频率响应音响系统的频率特性常用分贝刻度的纵坐标表示功率和用对数刻度的横坐标表示频率的频率响应曲线来描述。

频率响应是对MP3播放器的数模/模数转换器频率响应能力的一个评价标准。

好的频率响应,是在每一个频率点都能输出稳定足够的信号,不同频率点彼此之间的信号大小均一样。

然而在低频与高频部分,信号的重建比较困难,所以在这两个频段通常都会有衰减的现象。

输出品质越好的装置,频率响应曲线就越平直,反之不但在高低频处衰减得很快,在一般频段,也可能呈现抖动的现象。

频率响应是指将一个以恒电压输出的音频信号与系统相连接时,音箱产生的声压随频率的变化而发生增大或衰减、相位随频率而发生变化的现象,这种声压和相位与频率的相关联的变化关系(变化量)称为频率响应,频率响应范围是最低有效声音频率到最高有效声音频率之间的范围,单位为赫兹(Hz)THD+N 总谐波失真+噪声THD+N是英文Total Hormonic Distortion +Noise 的缩写译成中文是“总谐波失真加噪声”。

它是音频功率放大器的一个主要性能指标,也是音频功率放大器的额定输出功率的一个条件。

实际的音频功率放大器有各种谐波造成的失真及由器件内或外部造成的噪声,它有一定的THD+N的值。

这个值一般在0.00n%-10%之间(n=1~9)。

THD+N表示失真+噪声,因此THD+N自然越小越好。

但这个指标是在一定条件下测试的。

同一个音频功率放大器,若改变其条件,其THD+N的值会有很大的变动。

一般说,输出功率小(如几十mW)的高质量音频功率放大器(如用于MP3播放机),它的THD+N指标可达10-5,具有较高的保真度。

输出几百mW的音频功率放大器,要用扬声器放音,其THD+N一般为10-4;输出功率在1~2W,其THD+N更大些,一般为0.1~0.5%。

THD+N这一指标大小与音频功率放大器的结构类别有关(如A类功放、D类功放),例如D类功放的噪声较大,则THD+N的值也较A类大。

音频测试大全(AT相关仪器通用)

音频测试大全(AT相关仪器通用)

音频测试方法之迟辟智美创作Version: 1.52010-9-9DeclarationCircuit diagrams and other information relating to products of Actions Semiconductor Company, Ltd. (“Actions”) are included as a means of illustrating typical applications. Consequently, complete information sufficient for construction is not necessarily given. Although the information has been examined and is believed to be accurate, Actions makes no representations or warranties with respect to the accuracy or completeness of the contents of this publication and disclaims any responsibility for inaccuracies. Information in this document is provided solely to enable use of Actions’ products. The information presented in this document does not form part of any quotation or contract of sale. Actions assumes no liability whatsoever, including infringement of any patent or copyright, for sale and use of Actions’ products, except as expressed in Actions’ Terms and Conditions of Sale for. 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Any and all such uses without prior written approval of an Officer of Actions and further testing and/or modification will be fully at the risk of the customer. Copies of this document and/or other Actions product literature, as well as the Terms and Conditions of Sale Agreement, may be obtained by visiting Actions’ website at or from an authorized Actions representative. The word “ACTIONS”, the Actions’ LOGO, whether used separately and/or in combination, are trademarks of Actions Semiconductor Company, Ltd., Names and brands of other companies and their products that may from time to time descriptively appear in this product data sheet are the trademarks of their respective holders; no affiliation, authorization, or endorsement by such persons is claimed or implied except as may be expressly stated therein.ACTIONS DISCLAIMS AND EXCLUDES ANY AND ALL WARRANTIES, INCLUDING WITHOUT LIMITATION ANY AND ALL IMPLIED WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE, TITLE, AND AGAINST INFRINGEMENT AND THE LIKE, AND ANY AND ALL WARRANTIES ARISING FROM ANY COURSE OF DEALING OR USAGE OF TRADE.IN NO EVENT SHALL ACTIONS BE RELIABLE FOR ANY DIRECT, INCIDENTAL, INDIRECT, SPECIAL, PUNITIVE, OR CONSEQUENTIAL DAMAGES; OR FOR LOST DATA, PROFITS, SAVINGS OR REVENUES OF ANY KIND; REGARDLESS OF THE FORM OF ACTION, WHETHER BASED ON CONTRACT; TORT; NEGLIGENCE OF ACTIONS OR OTHERS; STRICT LIABILITY; BREACH OF WARRANTY; OR OTHERWISE; WHETHER OR NOT ANY REMEDY OF BUYER IS HELD TO HAVE FAILED OF ITS ESSENTIAL PURPOSE, AND WHETHER ACTIONS HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES OR NOT.Additional SupportAdditional product and company information can be obtainedby visiting the Actions website at:目录1适用范围52测试仪器与连接53测试目标参数64测试方法74.6 静态范围114.9 本底噪声扫描144.10 频响曲线164.12THD+N曲线184.13 SRS WOW曲线184.14 Line In184.16 FM录音205附录211、适用范围:本文档适用于媒体播放器样机的方案级音频部份的评估测试.2、测试仪器与连接音频分析仪:AP2700或AP2722电源: 干电池或锂电池 或精密电压源负载: 33欧姆/16欧姆/8欧姆电阻负载信号源: FM 信号发生器,AP2722或AP2700,MP3播放器+音箱耳机: 16欧姆/24欧姆/32欧姆3、 测试目标参数1、输出幅度(Max Amplitude ) 2、 输出功率(Output Power )(最年夜功率&最年夜待测样机 音频分析仪左右声道电缆 电阻负载PC电源不失真输出功率)3、总谐波失真加噪声(THD+N)4、地噪声(Ground Noise)5、信噪比(SNR)6、静态范围(Dynamic Range)7、串扰(Crosstalk)/ 通道分离度(ChannelSeparation)8、通道不服衡度9、本底噪声曲线(Ground Noise)10、频率响应曲线(Frequency Response)11、EQ曲线12、扫描THD+N曲线丈量13、SRS曲线丈量14、Line In 测试15、MIC录音测试16、FM录音测试17、主观听觉效果18、欧洲声压标准限制第二部份(EN50332 Limit Part2)4、测试方法描述1.输出幅度首先将待测样机设置为播放需要丈量频点的正弦波,比如1KHz或者100Hz,调节音量(测最年夜幅度时将音量调到最年夜;测100mV时调节音量使输出幅度最接近100mV),连接待测样机与音频分析仪,连接模拟电阻负载,依照上图设置AP2722的测试软件,待样机输出稳定后,切换声道,分别记录A、B通道的丈量数据.输出幅度一般需要测试100Hz,1KHz,10KHz,14KHz 和20KHz.2.输出功率首先将待测样机设置为播放1KHz正弦波,连接待测样机与音频分析仪,连接模拟电阻负载,依照上图进行输出功率丈量设置,从0级到最年夜,改变输出音量,切换声道,分别记录A、B通道的丈量数据.计算参考负载电阻依照实际连接的电阻负载进行设置,如果负载为16欧姆,则将Watts内的数值改为16欧姆,一般丈量中,负载为33欧姆.输出功率与输出幅度可以进行如下换算:(4-1)最年夜输出功率:当音量最年夜(THD+N不超越-60dB)时,输出的功率最年夜不失真输出功率:当某音量下,THD+N参数最好时,输出的功率3.总谐波失真加噪声首先将待测样机设置为播放待测频率的正弦波,连接待测样机与音频分析仪,连接负载电阻,并依照图4-3进行软件设置,调节音量,使得读取到的数据最好的情况下,切换声道,分别记录A、B通道的丈量数据.如需丈量A-Weight(A加权),使用AP2700仪器,在Fltr选择Slot 2# ,使用AP2722仪器,选择A-Weight.使用完成后,还原为NONE.4.Ground Noise(地噪声)连接测试设备,接好负载电阻.丈量最年夜地噪声时,将音量调到最年夜,播放0Hz文件,分别记录A、B声道的测试数据;丈量100mV的低噪声,先播放1KHz文件,调节音量使输出幅度最接近100mV,然后坚持此音量并播放0Hz文件,分别记录A、B声道的测试数据.如需丈量A-Weight(A加权),使用AP2700仪器,在Fltr选择Slot 2# ,使用AP2722仪器,选择A-Weight.使用完成后,还原为NONE.5.信噪比首先将待测样机设置为播放1KHz(或100Hz,10KHz,14KHz)的正弦波,调节音量(测最年夜幅度时将音量调到最年夜;测100mV时调节音量使输出幅度最接近100mV),连接待测样机与音频分析仪,连接负载电阻,并依照图4-4的左半部份进行软件设置,将输出数据设为相对值,按一下F4键,使得软件能够记录下参考信号幅度,此时丈量数据应为0dBrA左右.再播放0Hz文件,读取并记录相对数据.设置输出数据为绝对值,读取并记录绝对数据.切换到B通道,记录B通道的绝对数据,再将数值记录项改为dBrB,读取并记录相对数据.相对数据与绝对数据的关系如下:4-2)上式中,Vs为参考信号幅度,Vn为读取到的绝对数据.对干电池方案,需要丈量BAT端为最高电压和最低电压时的SNR.一般一节干电池方案,丈量电压为 1.5V和1.0V.如需丈量A-Weight(A加权),使用AP2700仪器,在Fltr选择Slot 2# ,使用AP2722仪器,选择A-Weight.使用完成后,还原为NONE.6.静态范围连接测试设备,接好负载电阻,设置音量,在最年夜音量下,播放目录\\srv-file\PUBFile\test-lib\Lib_AudioTestFiles\Standard_AudioTest_V1.1\MP3下的文件,丈量THD+N,获得的数据加上-40dB即为静态范围.记录数据.7.串扰丈量接好负载电阻,连接测试设备,设置样机在输出幅度为最接近100mV的音量下,分别播放左右单声道文件(3和,和,和,和),依照图4-5设置测试项为Crosstalk,调节通道,读取并记录数据.8.通道不服衡度通道不服衡度暗示左右通路输出幅度的分歧,依照上图读取并记录数据.丈量A通道选择A,丈量B通道选择B. 9.本底噪声曲线扫描(音频分析仪的FFT应用)连接测试设备,接好负载电阻,并依照以下步伐进行设置:按下Page2,选择扫描面板并在主菜单下选择Digital Analyzer(数字分析面板).呈现如下窗口:设置完成数字分析面板后,进行扫描设置:将纵坐标设置为FFT的幅度,将横坐标设置为FFT的频率,设置起止频率为音频范围,如20~22KHz,此设置为丈量输入信号的FFT分量.设置完成后,按下上图的GO按钮,开始进行分析,获得上图右边的黄色数据.此曲线即为以后时间点信号的频谱,由于丈量中使用音源为0Hz文件,即噪声文件,上图数据为样机的噪声平台(本底噪音).另外应用FFT还能够丈量谐波失真等用途.10.频响曲线丈量类似于FFT扫描,设置如下图的频率响应曲线扫描,完成后,样机在待测音量下播放文件,同时按下GO按钮,开始进行数据抓取.音源为开始和结束10秒为1KHz 正弦波,而后从20Hz开始,以1%的速率增长,直到22KHz,在输出频率为1KHz时,按下F4键,保管以后频响曲线的参考值(0dBr).测试音乐播放完后,扫描会自动停止.保管数据.一般频率响应曲线是在最年夜音量下丈量到的,因此,频率响应可以与最年夜输出幅度互相转换.频响曲线越平坦,对信号的还原度越好.11.EQ曲线丈量音效指MP3播放时,在解码中增加滤波器,使得输出的音频信号在某些频段上加强或者减弱,EQ曲线是丈量在各种音效下的频率响应曲线.播放文件为,首先将样机的输出音量调到最年夜,音效为Normal,依照频率响应曲线的设置方法进行设置,但要选择扫描设置窗口中的Append选项.在样机输出达到1KHz左右时,按下F4键保管以后参考值.接下来修改音效为Rock、Jazz……等,重新播放测试文件,获得一组曲线.修改线宽和颜色,并保管数据.每条曲线都以Normal在1KHz的值为参考0dB.对EQ,同时也需要丈量THD+N,测试方法详见下一小节.对每一种音效,都需要保管一条THD+N的曲线.EQ的THD+N扫描文件使用12.扫描THD+N曲线丈量分歧频率下,THD+N也不尽相同.类似于频率响应曲线和EQ曲线的丈量,在基本丈量面板上,修改丈量选项为THD+N Ratio.在扫描面板上,修改纵坐标为THD+N.播放文件,同时点击按钮“Go”,开始扫描,结束后,调整丈量范围,保证曲线有较年夜的分辨率,保管数据. 13.SRS WOW曲线丈量SRS WOW的丈量包括4个分量选项,SRS 3D,Focus,Trubass和默认SRSWOW(SRS 3D=8 ,Focus=8,Trubass=4);每一种分量下需要进行2个分歧文件的播放丈量,和(0相位和90相位).丈量SRS WOW曲线时,系统连接和设置与丈量频响曲线相同,选上扫描设置窗口中的Append选项,音量改为24级(为防止输出饱和).首先丈量SRS 3D,将SRS 3D 分量调节为10,将Focus和Trubass分量调节为0,播放F44.mp3文件,待输出频率达到1KHz时,按下F4,保管相对参考值.接下来需要丈量SRS 3D播放文件和剩下三个分量的6项丈量.以上七个丈量数据均以dBr为单元,绝对参考值为SRS 3D的0相位在1KHz的输出幅度.最后在一张图上保管8条曲线输出,即为SRS WOW的丈量结果.14.Line In丈量如果样机带有Line In接口,可以使用音频分析仪输出正弦波,通过样机录音,再进行播放,使用音频分析仪进行丈量.输出配置如下图,配置完成后,将分析仪的信号输出端口连接到样机的Line In接口,选择Line In,录制MP3文件,再回放,丈量SNR和THD+N.由于系统会对Line In信号进行放年夜,因此,可能发生饱和,在确定FW中设定的增益后,选择合适的输出幅度进行丈量.一般固件会将增益设置为0dB,可以采纳以下设置进行:对213X系列/22XX:Audio Precision的输出可以设置为2.6Vpp和0Vpp;对9X系列(不包括22XX):Audio Precision的输出可以设置为1.8Vpp和0Vpp.分别录音获得信号和噪声,输出丈量时不用带负载.需要时可用示波器观察输出波形.213X系列/22XX的正常指标约为:幅度490mV,THD+N-80dB,SNR-83dB.9X系列的正常指标约为:幅度275mV,THD+N-74dB,SNR-78dB.15.MIC录音测试使用声压计丈量布景噪声声压品级,设置为Lo量程,Fast模式,A加权.布景噪声需要小于60dB.使用MP3播放器播放1KHz歌曲,此播放器应保证此时输出的失真度不高于-60dB,将播放器连接到音箱上,如果是立体声音箱,需要去失落一个音箱,因为相干多音源在空间中会发生叠加,招致DUT处的声压过高或者过低.测试过程中,尽量坚持宁静.在MIC处,使用声压计丈量声压品级,设置为Hi量程,Fast模式,A加权.把持可以参考下图.调节音源音量,使得丈量点的声压品级为90dBA,在此条件下,DUT的MIC方向也应该对着音源方向,进行录音.录音结束后,将录音文件重放,并在音频分析仪上进行最年夜输出幅度和失真度的丈量,并记录数据. 16.FM录音测试使用FM信号发生器,在90MHz,98MHz和106MHz三个载波频点,具体可以《参考FM测试标准》,发生60dBuV的FM信号输出,调制信号为1KHz正弦波,在样机上进行录音并保管文件.在最年夜音量下,进行最年夜幅度、THD+N的丈量.17.主观听觉效果使用16欧姆、24欧姆和32欧姆负载耳机,在比力宁静的处所,播放静音文件,把持机器,包括改变音量,切换其他功能等,注意是否能听到细微的噪音,启动和停止、开机关机、Standby和唤醒过程是否存在POP音.播放单声道文件,是否串音明显.播放正常歌曲,在主观听觉上是否失真、声音是否清晰,音量变动是否均匀,EQ模式是否有效,Video播放,MIC录音,FM录音,Line In等是否存在人耳能够感知的失真等.18.欧洲声压标准限制第二部份(EN50332 Limit Part 2)依照丈量最年夜输出幅度,设置音频分析仪,播放文件Gauss_Noise_-10dB.MP3,丈量样机输出电压.调节音量,使得左右声道的输出幅度都刚刚低于150mV 时,将所有音效都丈量一遍,如果存在超越150mV,再减小音量,直到所有音效的输出幅度都低于150mV.记录以后输出幅度和音量品级.文件目录:\\srv-srd\Test-File\test-lib\Lib_AudioTestFiles\OtherTestFiles\Gauss_Noise5、附录1.丈量信号源文件路径(1)普通测试音频文件.测试文件可以在目录下找到.测试文件包括:NO1_16_200_44.MP3NO1_16_200_44.WMA NO2_16_15K_44.MP3NO2_16_15K_44. WMANO3_16_1K_44.MP3NO3_16_1K_44. WMANO4_16_0_44.MP3NO4_16_0_44. WMANO5_16_1KR_44.MP3NO5_16_1KR_44. WMANO6_16_1KL_44.MP3NO6_16_1KL_44. WMANO7_16__44.MP3NO7_16__44.WMANO8_16_127_44.MP3NO8_16_127_44.WMANO9_16_20K_44.MP3NO9_16_20K_4NO3_16_1K_44暗示编号3号的信号频率为1KHz,44.1K 采样率.注:测试WMA文件音频技术指标与测试MP3文件音频技术指标方法相同.数据记录在表格中有添加.(2)音效曲线丈量文件测试文件可以在目录下找到,包括:fr_-20db.OGGfr_-20db.wavfr_-20db.wma-20dB或者0dB暗示衰减为-20dB或者不衰减.(3)SRS评估音频文件测试文件可以在\\srv-file\PUBFile\test-lib\Lib_AudioTestFiles\Standard_Swept_V1.0\SRS扫频测试文件目录下找到.测试文件包括:(3)单声道测试文件:\\srv-file\PUBFile\test-lib\Lib_Music_Mp3\Lib_Mp3_1chmute2.名词解释SNR: 信噪比某个信道中,信号幅度与噪声幅度的比值SNR= -20log(V噪声/V信号)单元:dB“信号”丈量一般采纳的是指定输出电平的中频段正弦信号(通常为1kHz),由于丈量不成防止的会增加白噪声,这里的信号实际为信号+噪声,但正弦信号幅值远年夜于噪声幅度,在此将噪声忽略.“指定电平”通常是指设备的最年夜标称或标准的工作电平.“噪声”丈量必需指定丈量带宽和加权滤波器.两个丈量的比值就是设备的信噪比.因此,在参考信号幅度年夜小相同时,信噪比与本底噪声可以互相转换.由于分歧客户有分歧的标准,参考信号幅度年夜小是分歧的,因此,必需注意此时SNR的意义是否与客户端信噪比意义相同.另外,SD&Analog部份经常给出信噪比属于DAC+PA的信噪比,由于方案自己存在干扰,因此,方案丈量到的SNR不会超越SD部份给出的数据,但2者会比力接近. CROSSTALK: 分离度(串音,通道隔离度)双通道中,一个通道的输出信号电压与在另一个通道引起的信号电压之比.CROSSTALK= 20log(V1/V2) 单元:dBV1指无信号输出通道的电压均方根值,V2指有信号输出通道的电压均方根值,通常V2是1kHz正弦波发生的.串音一般是利用分析仪调谐到发生器频率的带通滤波器进行选择性丈量,以便能丈量即是或低于宽带噪声电平的串音.这不单是出于理论上的考虑原因;当信号的幅度处于宽带噪声电平之下10dB~20dB,人耳能够区分出象正弦波这样的相干信号.A通道到B通道的串音与B通道到A通道的串音其实不是完全一致的.两个方向上串音具有分歧值通常是电路的规划和复杂的寄生电容引起的.总谐波失真(THD)THD(不要与THD+N,总谐波失真加噪声相混淆)通常是由一系列独自谐波幅度丈量结果计算出来的,而不是一次丈量获得的.THD是独自谐波幅度的平方求和开方之后获得的.THD技术指标一般要说明包括在计算中的最高次谐波的次数;比如,“THD含盖到5次谐波”.THD其实不是经常进行的丈量,因为它要求用一个相当不经常使用的分析仪来丈量低于正常工作电平很多的某次谐波,而且要自动或手动计算出结果.应注意的是,许多早期的THD+N结构的分析仪在其面板上标注的是THD,而且许多人在使用的实际是THD+N技术时,认为是THD丈量.总谐波失真+噪声(THD+N)目前最经常使用的失真丈量方法就是THD+N技术了.其中的主要功能块就是可调谐的陷波器.在工作时,该滤波器手动或自动调谐到正弦波的基波频率上,以便基波被很年夜衰减.所设计的滤波器实际在2次和高次谐波处没有拔出损耗,所以谐波基本上无衰减地通过.宽带噪声,与AC电源有关的哼声和任何其他处在陷波器频率上下的干扰信号也可以无衰减地通过;这也就是“+N”(加噪声)部份的由来.THD+N技术是极为吸引人的,因为DUT输出中除纯丈量信号的任何成份城市使丈量下降.低的THD+N丈量结果不单说明谐波失真低,而且也说明哼声,干扰信号,以及宽带白噪声也是比丈量值低(或即是丈量值).所以THD+N比任何其他的失真丈量技术更能说明问题,它只用一个数据就能说明DUT是否存在年夜的问题静态范围:一个信号系统的静态范围被界说成最年夜不失真电平和噪声电平的差.而在实际用途中,多用对数和比值来暗示一个信号系统的静态范围,比如在音频工程中,一个放年夜器的静态范围可以暗示为:D = 10lg(Power_max / Power_min)=20 lg (Vout_Max/Vout_Min)使用数字量衰减的信号的目的是保证以后功率放年夜器没有包括自身存在的较年夜谐波失真.因而假设以后输出信号的THD+N丈量数值为Ground Noise的数值.此时的THD+N的分贝数值暗示Ground Noise与输出信号的比值,加上数字衰减失落的部份实际上获得的是Ground Noise与最年夜不失真输出信号的比值.SRS WOW技术SRS(Sound Retrieval System)是由SRS研究所开发的、最具代表性的3D立体声技术.该技术的核心是可以利用2个扬声器获得环绕立体声的效果.更具体一点说,声波在周边各空间点都有其特定的音量振幅、传布速度以及位置参数等,人的耳朵可以非常灵敏地分析这些数据,捕捉声音的强弱、移动方向以及位置. 此时,存在声音的滞后和破音(当声波碰到外耳之后呈现乱反射的现象),在普通扬声器中对声波延迟进行适当处置之后,再行输出的技术即为SRS技术.典范丈量单元意义说明:0dB: 指功放设备发出的电信号没有经过任何衰减的最年夜输出幅度.0dBr 参数:指相对参数的幅度(功率)的比值,其中参数为系统中的标定固定值.dBV: 指与1伏特(1V)信号比力得出的分贝数值.dBm: 指与1毫瓦(1mW)信号比力得出的分贝数值.dBu或者dBv: 指与0.774V(0.774V在600欧姆负载上发生1mW功率)信号比力得出的分贝数值.dB SPL(Sound Pressure Level):指声压1Pa(1牛顿/平方米)对应的品级.1Pa对应94 dB SPL.以上功率对数在分歧领域有分歧的默认值,如在通信领域内,有75欧姆和50欧姆等需要注意.AWeight Filter:在生理学中,人耳对应的频率响度曲线不是平坦的,而是符合Fletcher-Munson 的等响曲线( Equal Loudness Level Contours ),A Weight Filter 暗示基于40Phon的Fletcher-Munson曲线.声压品级(Sound Pressure Level):指声音能量在空气中传布,对接收装置的震动传布,单元是Pa.1Pa=1N/平方米=94dB SPL 其中,0dB SPL是指人的可闻阈值,其绝对值一般是0.00002帕斯卡(牛顿/平方米)欧盟标准EN50332:此标准一共有2个部份,第一部份规定了整机携带耳机的输作声压品级限制,不超越100dB SPL.第二部份规定了整机不携带耳机的输出电信号最高不超越150mV.对MP3/MP4产物来说,一般需要通过的是第一部份(声音信号),即声压品级不超越100dB SPL.而在实际研发生产中,绝年夜部份单元没有条件进行这一部份测试.因此,我们需要通过第二部份(电信号)即可,在考虑配备耳机时,选择合适灵敏度的耳机,即可通过第一部份.固然,以上数据都是在特殊的输出信号下丈量到的.标准规定,这个特殊信号的平均功率谱密度是平稳的,可以模拟音乐的.实际采纳高斯噪声,在数字媒体播放器中,数字量衰减10dB.实际丈量文件可以参考或者使用Gauss_Noise_-10dB.MP3文件.如果耳机的灵敏度是90dB/mW,其线圈电阻为32欧姆.则在输入为1000Hz信号,1mW功率时,发生的声压在人头模型中是90dB SPL.固然,由于耳机的频响不服坦,对存在于音频全频段的高斯噪声来说,标称灵敏度和实际丈量到的声压存在一定的误差,但一般丈量值会低于计算值.对输出电信号幅度的限制,可以通过播放Gauss_Noise_-10dB.MP3标准测试文件,在AP测试仪上,通过20Hz~22KHz的A-weight加权滤波器进行RMS幅度丈量,得出的数据即为标准中的输出电信号幅度.测试声压品级时,需要使用人头模型,模拟耳机对人耳的作用.如下图:测试文件提取的部份特征.高斯噪声时域波形.高斯噪声频谱分析.Version HistoryDate Revision Description11th June., 2007 初版正式发布.8th Dec.,2007 增加频响曲线的丈量,本底噪音频谱分析的丈量,增加典范单元的说明,修改失真度丈量部份的说明,增加信噪比部份的名词说明.15th Jan., 2008 1.2 增加Line In录音,FM录音部份测试说明,输出功率测试和静态范围测试.24th Apr., 2008增加MIC录音,测试干电池方案电压(1.5V和1.0V,使用精密电压源)对SNR的影响.增加31.5Hz,20KHz的输出幅度测试项目和20KHz失真度的测试项目.增加通道不服衡度的项目.修改静态范围曲目路径.增加WMA音源测试.25th June., 2008 增加A加权测试项目及说明增加THD+N的频率扫描项目及说明.增加EQ下的 THD+N曲线扫描说明.增加欧盟标准EN50332 Part2 测试方法及测试文件说明增加欧盟标准EN50332声压品级解释说明.15th Sept., 2010 更改测试文件:删除31.5Hz、200Hz、15KHz,增加100Hz、10KHz、14KHz增加测试100mV下输出幅度的说明增加测试100mV下THD+N的说明增加测试100mV下地噪声的说明增加测试100mV下SNR的说明增加测试100mV下crosstalk的说明。

音频测试大全(AT相关仪器通用)

音频测试大全(AT相关仪器通用)

音频测试办法Version: 1.52010-9-9DeclarationCircuit diagrams and other information relating to products of Actions Semiconductor Company, Ltd. (“Actions”) are included as a means of illustrating typical applications. Consequently, complete information sufficient for construction is not necessarily given. Although the information has been examined and is believed to be accurate, Actions makes no representations or warranties with respect to the accuracy or completeness of the contents of this publication and disclaims any responsibility for inaccuracies. Information in this document is provided solely to enable use of Actions’ products. The information presented in this document does not form part of any quotation or contract of sale. Actions assumes no liability whatsoever, including infringement of any patent or copyright, for sale and use of Actions’ products, except as expressed in Actions’ Terms and Conditions of Sale for. 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Additional SupportAdditional product and company information can be obtained by visiting the Actions website at:目次1实用规模52测试仪器与衔接53测试目标参数64测试办法74.6 动态规模114.9 本底噪声扫描144.10 频响曲线164.12THD+N曲线184.13 SRS WOW曲线184.14 Line In184.16 FM灌音205附录211、实用规模:本文档实用于媒体播放器样机的计划级音频部分的评估测试.2、测试仪器与衔接音频剖析仪:AP2700或AP2722电源:干电池或锂电池或周详电压源负载: 33欧姆/16欧姆/8欧姆电阻负载旌旗灯号源:FM旌旗灯号产生器,AP2722或AP2700,MP3播放器+音箱耳机: 16欧姆/24欧姆/32欧姆电源电阻负载阁下声道电缆待测样机音频剖析仪PC3、测试目标参数1、输出幅度(Max Amplitude)2、输出功率(Output Power)(最大功率&最大不掉真输出功率)3、总谐波掉真加噪声(THD+N)4、地噪声(Ground Noise)5、信噪比(SNR)6、动态规模(Dynamic Range)7、串扰(Crosstalk)/ 通道分别度(ChannelSeparation)8、通道不服衡度9、本底噪声曲线(Ground Noise)10、频率响应曲线(Frequency Response)11、EQ曲线12、扫描THD+N曲线测量13、SRS曲线测量14、Line In 测试15、MIC灌音测试16、FM灌音测试17、主不雅听觉后果18、欧洲声压尺度限制第二部分(EN50332 Limit Part2)4、测试办法描写1.输出幅度起首将待测样机设置为播放须要测量频点的正弦波,比方1KHz 或者100Hz,调节音量(测最大幅度时将音量调到最大;测100mV时调节音量使输出幅度最接近100mV),连招待测样机与音频剖析仪,衔接模仿电阻负载,按照上图设置AP2722的测试软件,待样机输出稳固后,切换声道,分别记载 A.B通道的测量数据.输出幅度一般须要测试100Hz,1KHz,10KHz,14KHz和20KHz.2.输出功率起首将待测样机设置为播放1KHz正弦波,连招待测样机与音频剖析仪,衔接模仿电阻负载,按照上图进行输出功率测量设置,从0级到最大,转变输出音量,切换声道,分别记载A.B通道的测量数据.盘算参考负载电阻按照现实衔接的电阻负载进行设置,假如负载为16欧姆,则将Watts内的数值改为16欧姆,一般测量中,负载为33欧姆.输出功率与输出幅度可以进行如下换算:(4-1)最大输出功率:当音量最大(THD+N不超出-60dB)时,输出的功率最大不掉真输出功率:当某音量下,THD+N参数最好时,输出的功率3.总谐波掉真加噪声起首将待测样机设置为播放待测频率的正弦波,连招待测样机与音频剖析仪,衔接负载电阻,并按照图4-3进行软件设置,调节音量,使得读取到的数据最好的情形下,切换声道,分别记载A.B通道的测量数据.如需测量A-Weight(A加权),运用AP2700仪器,在Fltr选择Slot 2# ,运用AP2722仪器,选择A-Weight.运用完成后,还原为NONE.4.Ground Noise(地噪声)衔接测试装备,接好负载电阻.测量最大地噪声时,将音量调到最大,播放0Hz文件,分别记载A.B声道的测试数据;测量100mV 的低噪声,先播放1KHz文件,调节音量使输出幅度最接近100mV,然后保持此音量并播放0Hz文件,分别记载A.B声道的测试数据.如需测量A-Weight(A加权),运用AP2700仪器,在Fltr选择Slot 2# ,运用AP2722仪器,选择A-Weight.运用完成后,还原为NONE.5.信噪比起首将待测样机设置为播放1KHz(或100Hz,10KHz,14KHz)的正弦波,调节音量(测最大幅度时将音量调到最大;测100mV时调节音量使输出幅度最接近100mV),连招待测样机与音频剖析仪,衔接负载电阻,并按照图4-4的左半部分进行软件设置,将输出数据设为相对值,按一下F4键,使得软件可以或许记载下参考旌旗灯号幅度,此时测量数据应为0dBrA阁下.再播放0Hz文件,读取并记载相对数据.设置输出数据为绝对值,读取并记载绝对数据.切换到B通道,记载B通道的绝对数据,再将数值记载项改为dBrB,读取并记载相对数据.相对数据与绝对数据的关系如下:4-2)上式中,Vs为参考旌旗灯号幅度,Vn为读取到的绝对数据.对于干电池计划,须要测量BAT端为最高电压和最低电压时的SNR.一般一节干电池计划,测量电压为1.5V和1.0V.如需测量A-Weight(A加权),运用AP2700仪器,在Fltr选择Slot 2# ,运用AP2722仪器,选择A-Weight.运用完成后,还原为NONE.6.动态规模衔接测试装备,接好负载电阻,设置音量,在最大音量下,播放目次\\srv-file\PUBFile\test-lib\Lib_AudioTestFiles\Standard_AudioTest_V1.1\MP3下的文件,测量THD+N,得到的数据加上-40dB即为动态规模.记载数据.7.串扰测量接好负载电阻,衔接测试装备,设置样机在输出幅度为最接近100mV的音量下,分别播放阁下单声道文件(3和,和,和,和),按照图4-5设置测试项为Crosstalk,调节通道,读取并记载数据.8.通道不服衡度通道不服衡度暗示阁下通路输出幅度的不同,按照上图读取并记载数据.测量A通道选择A,测量B通道选择B.9.本底噪声曲线扫描(音频剖析仪的FFT运用)衔接测试装备,接好负载电阻,并按照以下步调进行设置:按下Page2,选择扫描面板并在主菜单下选择Digital Analyzer(数字剖析面板).消失如下窗口:设置完成数字剖析面板后,进行扫描设置:将纵坐标设置为FFT的幅度,将横坐标设置为FFT的频率,设置起止频率为音频规模,如20~22KHz,此设置为测量输入旌旗灯号的FFT分量.设置完成后,按下上图的GO按钮,开端进行剖析,得到上图右边的黄色数据.此曲线即为当前时光点旌旗灯号的频谱,因为测量中运用音源为0Hz文件,即噪声文件,上图数据为样机的噪声平台(本底噪音).别的运用FFT还可以或许测量谐波掉真等用处.10.频响曲线测量相似于FFT扫描,设置如下图的频率响应曲线扫描,完成后,样机在待测音量下播放文件,同时按下GO按钮,开端进行数据抓取.音源为开端和停滞10秒为1KHz正弦波,尔后从20Hz开端,以1%的速度增长,直到22KHz,在输出频率为1KHz时,按下F4键,保管当前频响曲线的参考值(0dBr).测试音乐播放完后,扫描会主动停滞.保管数据.一般频率响应曲线是在最大音量下测量到的,是以,频率响应可以与最大输出幅度互相转换.频响曲线越平展,对于旌旗灯号的还原度越好.11.EQ曲线测量音效指MP3播放时,在解码中增长滤波器,使得输出的音频旌旗灯号在某些频段上增强或者削弱,EQ曲线是测量在各类音效下的频率响应曲线.播放文件为,起首将样机的输出音量调到最大,音效为Normal,按照频率响应曲线的设置办法进行设置,但要选择扫描设置窗口中的Append选项.在样机输出达到1KHz阁下时,按下F4键保管当前参考值.接下来修正音效为Rock.Jazz……等,从新播放测试文件,得到一组曲线.修正线宽和色彩,并保管数据.每条曲线都以Normal在1KHz的值为参考0dB.对于EQ,同时也须要测量THD+N,测试办法详见下一末节.对于每一种音效,都须要保管一条THD+N的曲线.EQ的THD+N扫描文件运用12.扫描THD+N曲线测量不合频率下,THD+N也不尽雷同.相似于频率响应曲线和EQ曲线的测量,在根本测量面板上,修正测量选项为THD+N Ratio.在扫描面板上,修正纵坐标为THD+N.播放文件,同时点击按钮“Go”,开端扫描,停滞后,调剂测量规模,包管曲线有较大的分辩率,保管数据.13.SRS WOW曲线测量SRS WOW的测量包含4个分量选项,SRS 3D,Focus,Trubass和默认SRSWOW(SRS 3D=8 ,Focus=8,Trubass=4);每一种分量下须要进行2个不合文件的播放测量,和(0相位和90相位).测量SRS WOW曲线时,体系衔接和设置与测量频响曲线雷同,选上扫描设置窗口中的Append选项,音量改为24级(为防止输出饱和).起首测量SRS 3D,将SRS 3D分量调节为10,将Focus 和Trubass分量调节为0,播放F44.mp3文件,待输出频率到达1KHz时,按下F4,保管相对参考值.接下来须要测量SRS 3D播放文件和剩下三个分量的6项测量.以上七个测量数据均以dBr 为单位,绝对参考值为SRS 3D的0相位在1KHz的输出幅度.最后在一张图上保管8条曲线输出,即为SRS WOW的测量成果. 14.Line In测量假如样机带有Line In接口,可以运用音频剖析仪输出正弦波,经由过程样机灌音,再进行播放,运用音频剖析仪进行测量.输出设置装备摆设如下图,设置装备摆设完成后,将剖析仪的旌旗灯号输出端口衔接到样机的Line In接口,选择Line In,录制MP3文件,再回放,测量SNR和THD+N.因为体系会对Line In旌旗灯号进行放大,是以,可能产生饱和,在肯定FW中设定的增益后,选择适合的输出幅度进行测量.一般固件会将增益设置为0dB,可以采取以下设置进行:对于213X系列/22XX:Audio Precision的输出可以设置为2.6Vpp和0Vpp;对于9X系列(不包含22XX):Audio Precision 的输出可以设置为1.8Vpp和0Vpp.分别灌音得到旌旗灯号和噪声,输出测量时不必带负载.须要时可用示波器不雅察输出波形.213X系列/22XX的正常指标约为:幅度490mV,THD+N-80dB,SNR-83dB.9X系列的正常指标约为:幅度275mV,THD+N-74dB,SNR-78dB.15.MIC灌音测试运用声压计测量布景噪声声压等级,设置为Lo量程,Fast模式,A加权.布景噪声须要小于60dB.运用MP3播放器播放1KHz歌曲,此播放器应包管此时输出的掉真度不高于-60dB,将播放器衔接到音箱上,假如是立体声音箱,须要去掉落一个音箱,因为相关多音源在空间中会产生叠加,导致DUT处的声压过高或者过低.测试进程中,尽量保持安静.在MIC处,运用声压计测量声压等级,设置为Hi量程,Fast模式,A加权.操纵可以参考下图.调节音源音量,使得测量点的声压等级为90dBA,在此前提下,DUT的MIC偏向也应当对着音源偏向,进行灌音.灌音停滞后,将灌音文件重放,并在音频剖析仪长进行最大输出幅度和掉真度的测量,并记载数据.16.FM灌音测试运用FM旌旗灯号产生器,在90MHz,98MHz和106MHz三个载波频点,具体可以《参考FM测试尺度》,产生60dBuV的FM旌旗灯号输出,调制旌旗灯号为1KHz正弦波,在样机长进行灌音并保管文件.在最大音量下,进行最大幅度.THD+N的测量.17.主不雅听觉后果运用16欧姆.24欧姆和32欧姆负载耳机,在比较安静的地方,播放静音文件,操纵机械,包含转变音量,切换其他功效等,留意是否能听到细微的噪音,启动和停滞.开机关机.Standby和叫醒进程是否消失POP音.播放单声道文件,是否串音显著.播放正常歌曲,在主不雅听觉上是否掉真.声音是否清楚,音量变更是否平均,EQ模式是否有用,Video播放,MIC灌音,FM灌音,Line In等是否消失人耳可以或许感知的掉真等.18.欧洲声压尺度限制第二部分(EN50332 Limit Part 2)按照测量最大输出幅度,设置音频剖析仪,播放文件Gauss_Noise_-10dB.MP3,测量样机输出电压.调节音量,使得阁下声道的输出幅度都方才低于150mV时,将所有音效都测量一遍,假如消失超出150mV,再减小音量,直到所有音效的输出幅度都低于150mV.记载当前输出幅度和音量等级.文件目次:\\srv-srd\Test-File\test-lib\Lib_AudioTestFiles\OtherTestFiles\Gauss_Noise5、附录1.测量旌旗灯号源文件路径(1)通俗测试音频文件.测试文件可以在目次下找到.测试文件包含:NO1_16_200_44.MP3NO1_16_200_44.WMANO2_16_15K_44.MP3NO2_16_15K_44. WMANO3_16_1K_44.MP3NO3_16_1K_44. WMANO4_16_0_44.MP3NO4_16_0_44. WMANO5_16_1KR_44.MP3NO5_16_1KR_44. WMANO6_16_1KL_44.MP3NO6_16_1KL_44. WMANO7_16__44.MP3NO7_16__44.WMANO8_16_127_44.MP3NO8_16_127_44.WMANO9_16_20K_44.MP3NO9_16_20K_4NO3_16_1K_44暗示编号3号的旌旗灯号频率为1KHz,44.1K 采样率.注:测试WMA文件音频技巧指标与测试MP3文件音频技巧指标办法雷同.数据记载在表格中有添加.(2)音效曲线测量文件测试文件可以在目次下找到,包含:fr_-20db.OGGfr_-20db.wavfr_-20db.wma-20dB或者0dB暗示衰减为-20dB或者不衰减.(3)SRS评估音频文件测试文件可以在\\srv-file\PUBFile\test-lib\Lib_AudioTestFiles\Standard_Swept_V1.0\SRS扫频测试文件目次下找到.测试文件包含:(3)单声道测试文件:\\srv-file\PUBFile\test-lib\Lib_Music_Mp3\Lib_Mp3_1chmute2.名词解释SNR: 信噪比某个信道中,旌旗灯号幅度与噪声幅度的比值SNR= -20log(V噪声/V旌旗灯号)单位:dB“旌旗灯号”测量一般采取的是指定输出电平的中频段正弦旌旗灯号(平日为1kHz),因为测量不成防止的会增长白噪声,这里的旌旗灯号现实为旌旗灯号+噪声,但正弦旌旗灯号幅值弘远于噪声幅度,在此将噪声疏忽.“指定电平”平日是指装备的最大标称或尺度的工作电平.“噪声”测量必须指定测量带宽和加权滤波器.两个测量的比值就是装备的信噪比.是以,在参考旌旗灯号幅度大小雷同时,信噪比与本底噪声可以互相转换.因为不合客户有不合的尺度,参考旌旗灯号幅度大小是不合的,是以,必须留意此时SNR的意义是否与客户端信噪比意义雷同.别的,SD&Analog部分经常给出信噪比属于DAC+PA的信噪比,因为计划本身消失干扰,是以,计划测量到的SNR不会超出SD部分给出的数据,但2者会比较接近.CROSSTALK: 分别度(串音,通道隔离度)双通道中,一个通道的输出旌旗灯号电压与在另一个通道引起的旌旗灯号电压之比.CROSSTALK= 20log(V1/V2) 单位:dBV1指无旌旗灯号输出通道的电压均方根值,V2指有旌旗灯号输出通道的电压均方根值,平日V2是1kHz正弦波产生的.串音一般是运用剖析仪调谐到产生器频率的带通滤波器进行选择性测量,以便能测量等于或低于宽带噪声电平的串音.这不但是出于理论上的斟酌原因;当旌旗灯号的幅度处于宽带噪声电平之下10dB~20dB,人耳可以或许区分出象正弦波如许的相关旌旗灯号.A通道到B通道的串音与B通道到A通道的串音其实不是完整一致的.两个偏向上串音具有不合值平日是电路的计划和庞杂的寄生电容引起的.总谐波掉真(THD)THD(不要与THD+N,总谐波掉真加噪声相混杂)平日是由一系列单独谐波幅度测量成果盘算出来的,而不是一次测量得到的.THD是单独谐波幅度的平方乞降开方之后得到的.THD技巧指标一般要解释包含在盘算中的最高次谐波的次数;比方,“THD含盖到5次谐波”.THD其实不是经常进行的测量,因为它请求用一个相当不经常运用的剖析仪来测量低于正常工作电平很多的某次谐波,并且要主动或手动盘算出成果.应留意的是,很多早期的THD+N构造的剖析仪在其面板上标注的是THD,并且很多人在运用的现实是THD+N技巧时,以为是THD测量.总谐波掉真+噪声(THD+N)今朝最经常运用的掉真测量办法就是THD+N技巧了.个中的重要功效块就是可调谐的陷波器.在工作时,该滤波器手动或主动调谐到正弦波的基波频率上,以便基波被很大衰减.所设计的滤波器现实在2次和高次谐波处没有拔出损耗,所以谐波根本上无衰减地经由过程.宽带噪声,与AC电源有关的哼声和任何其他处在陷波器频率高低的干扰旌旗灯号也可以无衰减地经由过程;这也就是“+N”(加噪声)部分的由来.THD+N技巧是极为吸惹人的,因为DUT输出中除了纯测量旌旗灯号的任何成分都邑使测量降低.低的THD+N测量成果不但解释谐波掉真低,并且也解释哼声,干扰旌旗灯号,以及宽带白噪声也是比测量值低(或等于测量值).所以THD+N比任何其他的掉真测量技巧更能解释问题,它只用一个数据就能解释DUT是否消失大的问题动态规模:一个旌旗灯号体系的动态规模被界说成最大不掉真电温和噪声电平的差.而在现实用处中,多用对数和比值来暗示一个旌旗灯号体系的动态规模,比方在音频工程中,一个放大器的动态规模可以暗示为:D = 10lg(Power_max / Power_min)=20 lg (Vout_Max/Vout_Min)运用数字量衰减的旌旗灯号的目标是包管当前功率放大器没有包含自身消失的较大谐波掉真.因而假设当前输出旌旗灯号的THD+N测量数值为Ground Noise的数值.此时的THD+N的分贝数值暗示Ground Noise与输出旌旗灯号的比值,加上数字衰减掉落的部分现实上得到的是Ground Noise与最大不掉真输出旌旗灯号的比值.SRS WOW技巧SRS(Sound Retrieval System)是由SRS研讨所开辟的.最具代表性的3D立体声技巧.该技巧的焦点是可以运用2个扬声器获得围绕立体声的后果.更具体一点说,声波在周边各空间点都有其特定的音量振幅.传播速度以及地位参数等,人的耳朵可以异常敏锐地剖析这些数据,捕获声音的强弱.移动偏向以及地位.此时,消失声音的滞后和破音(当声波碰着外耳之后消失乱反射的现象),在通俗扬声器中对声波延迟进行恰当处理之后,再行输出的技巧即为SRS技巧.典范测量单位意义解释:0dB: 指功放装备发出的电旌旗灯号没有经由任何衰减的最大输出幅度.0dBr 参数:指相对于参数的幅度(功率)的比值,个中参数为体系中的标定固定值.dBV: 指与1伏特(1V)旌旗灯号比较得出的分贝数值.dBm: 指与1毫瓦(1mW)旌旗灯号比较得出的分贝数值.dBu或者dBv: 指与0.774V(0.774V在600欧姆负载上产生1mW功率)旌旗灯号比较得出的分贝数值.dB SPL(Sound Pressure Level):指声压1Pa(1牛顿/平方米)对应的等级.1Pa对应94 dB SPL.以上功率对数在不合范畴有不合的默认值,如在通讯范畴内,有75欧姆和50欧姆等须要留意.AWeight Filter:在心理学中,人耳对应的频率响度曲线不是平展的,而是相符Fletcher-Munson 的等响曲线( Equal Loudness Level Contours ),A Weight Filter 暗示基于40Phon的Fletcher-Munson曲线.声压等级(Sound Pressure Level):指声音能量在空气中传播,对于吸收装配的震撼传播,单位是Pa.1Pa=1N/平方米=94dB SPL 个中,0dB SPL是指人的可闻阈值,其绝对值一般是0.00002帕斯卡(牛顿/平方米)欧盟尺度EN50332:此尺度一共有2个部分,第一部分划定了整机携带耳机的输出声压等级限制,不超出100dB SPL.第二部分划定了整机不携带耳机的输出电旌旗灯号最高不超出150mV.对于MP3/MP4产品来说,一般须要经由过程的是第一部分(声音旌旗灯号),即声压等级不超出100dB SPL.而在现实研产临盆中,绝大部分单位没有前提进行这一部分测试.是以,我们须要经由过程第二部分(电旌旗灯号)即可,在斟酌配备耳机时,选择适合敏锐度的耳机,即可经由过程第一部分.当然,以上数据都是在特别的输出旌旗灯号下测量到的.尺度划定,这个特别旌旗灯号的平均功率谱密度是安稳的,可以模仿音乐的.现实采取高斯噪声,在数字媒体播放器中,数字量衰减10dB.现实测量文件可以参考或者运用Gauss_Noise_-10dB.MP3文件.假如耳机的敏锐度是90dB/mW,其线圈电阻为32欧姆.则在输入为1000Hz旌旗灯号,1mW功率时,产生的声压在人头模子中是90dB SPL.当然,因为耳机的频响不服坦,对消失于音频全频段的高斯噪声来说,标称敏锐度和现实测量到的声压消失必定的误差,但一般测量值会低于盘算值.对于输出电旌旗灯号幅度的限制,可以经由过程播放Gauss_Noise_-10dB.MP3尺度测试文件,在AP测试仪上,经由过程20Hz~22KHz的A-weight加权滤波器进行RMS幅度测量,得出的数据即为尺度中的输出电旌旗灯号幅度.测试声压等级时,须要运用人头模子,模仿耳机对人耳的感化.如下图:测试文件提取的部分特点.高斯噪声时域波形.高斯噪声频谱剖析.Version HistoryDate Revision Description11th June., 2007 第一版正式宣布.8th Dec.,2007 增长频响曲线的测量,本底噪音频谱剖析的测量,增长典范单位的解释,修正掉真度测量部分的解释,增长信噪比部分的名词解释.15th Jan., 2008 1.2 增长Line In灌音,FM灌音部分测试解释,输出功率测试和动态规模测试.24th Apr., 2008增长MIC灌音,测试干电池计划电压(1.5V和1.0V,运用周详电压源)对SNR的影响.增长31.5Hz,20KHz的输出幅度测试项目和20KHz掉真度的测试项目.增长通道不服衡度的项目.修修正态规模曲目路径.增长WMA音源测试.25th June., 2008 增长A加权测试项目及解释增长THD+N的频率扫描项目及解释.增长EQ下的 THD+N曲线扫描解释.增长欧盟尺度EN50332 Part2 测试办法及测试文件解释增长欧盟尺度EN50332声压等级解释解释.15th Sept., 2010 更改测试文件:删除31.5Hz.200Hz.15KHz,增长100Hz.10KHz.14KHz增长测试100mV下输出幅度的解释增长测试100mV下THD+N的解释增长测试100mV下地噪声的解释增长测试100mV下SNR的解释增长测试100mV下crosstalk的解释。

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模拟音频参数和测试1. 基本单位和概念dBu 以0.775V (有效值)为基准电压时的电压电平单位。

表示为:dBu=20lg(v/0.775)dBu 的计算只考虑电压电平本身,而不考虑与相应的电功率电平之间的关系,不考虑阻抗是否为600Ω。

---参照GY/T 192-2003dBu 采用接近0的源阻抗和接近无穷大的负载阻抗!基准信号的电平为0.775V RMS dBu=20log(Vx/0.775),Vx=0.775*10(Dbu/20),0.775V RMS 对应的电阻为600ohm,即1mW 在600ohm 产生0.77459的电压220.7750.001600U P WR ==注意dBu 表示的是电压值,在音频上并不是以1uV 作为基准电压,而是以0.775V RMS 作为基准 所以dBu 换算出来是RMS 值Vrms有效值,均方根值,正弦波时,均方根值Vrms为振幅Vm的0.707倍,为峰峰值的0.707/2倍Vpp峰峰值Vpp=2*Vm=2.828VrmsdBFs(dB below digital Full Scale)以满刻度的量值为0dB,常用于各种特性曲线上;数字音频信号测量中经常使用到单位“dbFS”。

0dbFS既是指满刻度的数字音频参考电平,即“数字满刻度电平”,它是指在数字域的音频系统中,A/D或D/A转换器可能达到的“数字过载”之前的最大可编码模拟信号电平。

0dbFS为数字音频信号最高峰的绝对值,与16bit线性编码PCM信号对应的最高值为7FFF(16进制),最高负值电平为8000(16进制),十进制数为32767。

不同国家对数字设备满度电平值OdBFS所对应的电平模拟信号的电平值不尽相同,目前还没有这个标准数字码的国际标准,常见的是SMPTE。

(美国电影电视工程师学会)和EBU(欧洲广播联盟)推荐的两个方案。

SMPTE推荐的转换基准规定为对于16bit的PCM声音信号,频率为lkHz的模拟正弦波信号的正、负峰值使A/D转换器分别产生OCCD,F333数字码时的幅度为参考电平。

OCCD和F333对应的十进制数为3277,因20lg3277/32767≈-20dBFS,所以SMPTE推荐的参考电平为-20dBFS。

EBU推荐的转换基准规定对于l6bit的PCM 声音信号,频率为lkHz的模拟正弦波信号的正、负峰值使A/D转换器分别产生0FFF,F000数字码时的幅度为参考电平。

0FFF,F000对应的十进制数为4095,因20lg4095/32767≈-18dBFS,所以EBU推荐的参考电平为-18dBFS。

针对不同的模拟基准电平,0dBFS所对应的模拟信号电压电平也不同。

由于中国广播系统中采用+4dBu 作为音频系统的校准电平,所以广播电影电视行业标准GY/T192规定数字设备的满度电平值0dBFS对应的模拟信号电平为24dBu,考虑到中国广播电台的实际情况,现阶段允许满度电平值0dBFS对应的模拟信号电压电平+22dBu的数字设备继续使用。

测试数字电视接收产品的音频输出电平时必须对测试码流中的音频信号电平进行规定。

GY/T-192规定中国的数宇音频满刻度电平0dBFS对应的模拟信号电平为+24dBu,但国际上对这个对应关系并没有统一的标准。

另外因应用场合不同,各种仪器设备的数字满引度所对应的模拟电平也不相同。

目前中国生产企业和测试机构多选用国外生产的数宇测试信号发生器,主要产品有美国泰克公司生产的MTG系列和R&S公司生产的DVRG等。

其中DVRG对音频信号有如下规定:0dBr=+6dBu=l.66V(DIN45406),0dBFS=满刻度,16bit编码,对应信号峰一峰的十进制数为65536;0dBr=- 6dBFS,16bit编码,对应信号峰一峰的十进制数为32768(土16384)。

可以看出,DVRG的数宇满刻度电平OdBFS对应的模拟电平不是+24dBu,而是+l2dBu,其参考电平是-6dBFS(+6dBu)。

《有线数字电视系统用户终端接收机入网技术条件和测量方法第一部分:透明传输电性能参数》(暂行)中规定:在测量数字有线电视接收机的音频输出电平时应采用-20dBFS(+4dBu)的信号,要求接收机的输出电平不小于-8dBu。

但是如果使用DVRG作为信号源,其-20dBFS所对应的不是+4dBu,而是-8dBu,如果仍采用-20dBFS的信号进行测试,会造成测试结果的不正确。

因此在使用数宇测试信号发生器前,应对其音频数字满刻度所对应的模拟信号电平进行确认,采用标准规定的信号电平进行测试,才能保证测试结果的准确性。

dBu和dBFS是有对应关系的FS Full Scale在美国ATSC系统中,0dBFS被定义为相当于模拟电平的24dBu(12.3V),在中国及欧洲备注对于16bit采样音频信号的动态范围是96DBdBFS = 20 * log (采样信号 / 1111 1111 1111 1111)20 * log (1111 1111 1111 1111 / 1111 1111 1111 1111) = 0 dBFS20 * log (0000 0000 0000 0001 / 1111 1111 1111 1111) = -96 dBFS(换成10进制)而24bit采样的采样音频信号的动态范围144.4943974 DB音频信噪比音频信噪比是指音响设备播放时,正常声音信号强度与噪声信号强度的比值。

当信噪比低,小信号输入时噪音严重,在整个音域的声音明显变得浑浊不清,不知发的是什么音,严重影响音质。

信噪比的大小是用有用信号功率(或电压)和噪声功率(或电压)比值的对数来表示的。

这样计算出来的单位称为“贝尔”。

实用中因为贝尔这个单位太大,所以用它的十分之一做计算单位,称为“分贝”。

对于便携式DVD来说,信噪比至少应该在70dB(分贝)以上,才可以考虑。

信噪比,即SNR(Signal to Noise Ratio),又称为讯噪比。

狭义来讲是指放大器的输出信号的电压与同时输出的噪声电压的比,常常用分贝数表示,设备的信噪比越高表明它产生的杂音越少。

一般来说,信噪比越大,说明混在信号里的噪声越小,声音回放的音质量越高,否则相反。

信噪比一般不应该低于70dB,高保真音箱的信噪比应达到110dB以上。

信噪比的测量及计算通过计算公式我们发现,信噪比不是一个固定的数值,它应该随着输入信号的变化而变化,如果噪声固定的话,显然输入信号的幅度越高信噪比就越高。

显然,这种变化着的参数是不能用来作为一个衡量标准的,要想让它成为一种衡量标准,就必须使它成为一个定值。

于是,作为器材设备的一个参数,信噪比被定义为了“在设备最大不失真输出功率下信号与噪声的比率”,这样,所有设备的信噪比指标的测量方式就被统一起来,大家可以在同一种测量条件下进行比较了。

信噪比通常不是直接进行测量的,而是通过测量噪声信号的幅度换算出来的,通常的方法是:给放大器一个标准信号,通常是0.775Vrms或2Vp-p@1kHz,调整放大器的放大倍数使其达到最大不失真输出功率或幅度(失真的范围由厂家决定,通常是10%,也有1%),记下此时放大器的输出幅Vs,然后撤除输入信号,测量此时出现在输出端的噪声电压,记为Vn,再根据SNR=20LG(Vn/Vs)就可以计算出信噪比了。

Ps和Pn 分别是信号和噪声的有效功率,根据SNR=10LG(Ps/Pn)也可以计算出信号比。

这样的测量方式完全可以体现设备的性能了。

但是,实践中发现,这种测量方式很多时候会出现误差,某些信噪比测量指标高的放大器,实际听起来噪声比指标低的放大器还要大。

经过研究发现,这不是测量方法本身的错误,而是这种测量方法没有考虑到人的耳朵对于不同频率的声音敏感性是不同的,同样多的噪声,如果都是集中在几百到几千Hz,和集中在20KHz以上是完全不同的效果,后者我们可能根本就察觉不到。

因此就引入了一个“权”的概念。

这是一个统计学上的概念,它的核心思想是,在进行统计的时候,应该将有效的、有用的数据进行保留,而无效和无用的数据应该尽量排除,使得统计结果接近最准确,每个统计数据都由一个“权”,“权”越高越有用,“权”越低就越无用,毫无用处的数据的“权”为0。

于是,经过一系列测试和研究,科学家们找到了一条“通用等响度曲线”,这个曲线代表的是人耳对于不同频率的声音的灵敏度的差异,将这个曲线引入信噪比计算方法后,先兆比指标就和人耳感受的结果更为接近了。

噪声中对人耳影响最大的频段“权”最高,而人耳根本听不到的频段的“权”为0。

这种计算方式被称为“A 计权”,已经称为音响行业中普遍采用的计算方式。

总谐波失真(THD )信号的失真情况,通常使用THD 也就是总谐波失真来表示,总谐波失真是指用信号源输入时,输出信号比输入信号多出的额外谐波成分。

谐波失真是由于系统不是完全线性造成的,它通常用百分数来表示,也可以用dB 来表示。

在正常工作的情况下,输出信号中总的谐波电压有效值与总输出信号的电压有效值之比。

所有附加谐波电平之和称为总谐波失真。

一般说来,1KHz 频率处的总谐波失真最小,因此不少产品均以该频率的失真作为它的指标。

但总谐波失真与频率有关,必须在20-20000Hz 的全音频范围内测出。

一般我们测试时测试THD+N(总谐波失真加噪声)左右声道串扰在多通道的放大器中,一个通道的信号可能会以衰减或失真的形式串进另一个通道,音频左右声道串扰是指当一个声道输入信号的时候,在另一个声道因为串过去的干扰所产生的信号强度,称为串扰,以dB 为单位(其实是dBr ),通常以0dB 单左声道和0dB 单右声道的音源测试L 声道对R 声道的串扰()20log ()L LR L U U左右声道相位差两个声道输入同一频率的信号时,由电路延时差异造成的相位差别(电容,电感都会造成相位差),通常以1KHz 为标准,所测值是个相位。

单位为度或者弧度左右声道电平差音频左右声道电平差,就是当两个声道输入同一幅度的信号时,输出部分由于电路增益差异造成的输出电平差别,以dB 为单位。

动态范围(dynamic range )数字音频的分辨率采样率44.1K 48K2. 几个测试指标(几个不同的标准,测试结果请注意单位)序号项目单位广电DVB-C要求IPTV要求备注1 音频输出电平dBu ≥-8 ≥-8 负载阻抗600测试信号为1KHZ/-20dBFs正弦波音频信号2 音频失真度%≤1.5 测试信号电平为1KHZ/-8dBFs测试频率范围为1KHZ3 音频幅频特性dB +1/-2 测试信号电平为-20dBFs测试频率范围为60HZ-18KHZ4 音频信噪比(不加权)dB ≥705 音频左右声道串扰dB ≤-706 音频左右声道相位差°≤57 音频左右声道电平差dB ≤0.5高清\标清机顶盒设备技术规范、待测项标准值测量值(1台样机)1 音频输出电平RMS(V) L 2.0±0.1 2.071V R 2.0±0.1 2.081V2 音频幅度响应(dB) L 20Hz-20KHz ±2 -3---+0.6(20Hz-20KHz)40Hz-18KHz ±1 -1---+0.6(40Hz-18KHz)R 20Hz-20KHz ±2 -3---+0.6(20Hz-20KHz)40Hz-18KHz ±1 -1---+0.6(40Hz-18KHz)3 谐波失真+噪声L ≤-65-60dB(0.09%)3.测试方法及所需要用到的片源0dB; 997Hz; Stereo(test1,2,6,7)0dB-20Hz-20000Hz-3s-67steps(test3)MP3 Test Tones - Infinity Zero(test4)0dB; 997Hz; Left(test5)0dB; 997Hz; Right(test5)4.几个音频相关的软件Ap2700专业音频编辑软件AdobeAuditionV3.0音频编辑软件GoldWavev5.52汉化绿色增强版5.音频部分设计中需要注意的几个问题音频电源噪声和滤波消除开关机POP元件品质对音频输出参数的影响。

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