模拟音频参数和测试
音响师的两款流行音频测量软件
音响师的两款流行音频测量软件作为专业音响师，如果不会那么几样音频测量软件，身边没有几套测试工具，都不好意思出去跟人家打招呼，顶多算个干音响的。
今天为大家介绍两款业内比较流行的声场测量工具：Smaartlive和EASERA SysTune。
音响师必备音频测量工具之一：EASERA SysTuneEASERA SysTune由四个模块组成：测量模块、信号发生器、实时分析器和后处理模块。
尤其可以使用swept sine和MLS来执行测量。
有了这些测量，就可以顾及室内及各种不同类别的背境噪声，这程序也可以分析wave 档，并可利用声卡的输入来记录信号。
声卡的校准很简单，只要卡上设有同步时钟，如果声卡有多个输入或输出，可用第二输入或输出作为同步和测定延迟，所有设定皆是易用的，尽管提供有大量的选择。
EASERA SysTune亦有自动化功能，比如话筒校准，调节输入灵敏度和补偿硬件的频率特性，可以在实时进行综合性的FFT测量，查阅音量表和显示谱图。
一切的预设皆可储存起来，以至可以再次使用，或者稍后在后处理期间使用。
真正的乐趣是在后处理上开始，赋与你计算Room Acoustics Electro Acoustic、RMS、噪声电平，回波图等等。
在这里以时域和频域操控各种信号也是可以的。
EASERA可用192KHz取样频率监视多至32个输入，这对于在音乐会中PA应用系统来讲是甚有意义的。
对于专业音响工程师来说EASERA是一套完美的工具，其功能实际是无穷尽的，可靠性是甚高的，除了可用SysTune以192KHz取样频率，在同时和在实时测量8条通道上的信号，还提供以下的功能：程序给于直接输入信号或内部产生信号的选择、脉冲响应测量，谱图测量、多通道FFT、在实时相位测量，以及更多。
不单给你对于输入信号进行各类的测量，而且还可做空间测量，以及在录音室和演奏厅作扬声器定位测量。
SysTune收购脉冲响应数据材料使用现场直播节目，并同时显示时域和频域的措施进行实时并自动产生精确的，可重复的测量每一次!SysTune也有能力处理多达8个输入通道的同时，创造空间和转让平均频谱显示功能。
recorder的sampledata方法
recorder的sampledata方法1.引言在软件开发中，我们经常需要模拟和测试数据，以验证我们的代码是否正确。
对于手机录音应用r ec or de r来说，sa mp le da ta方法是一个非常有用的功能，它可以生成模拟的音频数据。
本文将介绍r eco r de r的s a mp le da ta方法及其使用方式。
2. sa mpledat a方法的概述s a mp le da ta方法是r e co rd er库中的一个函数，用于生成模拟的音频数据。
它可以根据指定的参数生成不同类型的音频样本，如白噪音、音乐片段等，以供开发者进行测试和调试。
3.使用方法3.1安装r e c o r d er库首先，我们需要安装r ec or de r库。
使用以下命令，在你的项目中添加r ec or de r库的依赖：$n pm in st al lr ec ord e r--s av e3.2引入r e c o r d er库在你的代码中引入re c or de r库：c o ns tR ec or de r=req u ir e('r ec or de r');3.3调用s a m p l e d a t a方法在你的代码中使用sa m pl ed at a方法，生成模拟音频数据：c o ns ts am pl eD at a=R e co rd er.s am pl eda t a(op ti on s);3.4参数说明s a mp le da ta方法接收一个包含配置参数的o pt io ns对象，以指定音频数据的类型和属性。
以下是常用的参数说明：-`ty pe`（可选）：指定音频数据的类型，可以是白噪音、音乐片段等。
默认为白噪音。
-`du ra ti on`（可选）：指定音频数据的持续时间，单位为秒。
默认为10秒。
-`fr eq ue nc y`（可选）：指定音频数据的采样频率，单位为H z。
音频、FM测试参数及单位说明
P 0.001W
P
dBm
0.00110 10
2、dBu（欧洲用）和dBV（美国用） u和V是小写,负载情况不清楚。
dBv=dBu=20log10
U 0.775V
3、dBV V是大写,负载情况不清楚。
dBV=20log10
U 1V
3、dBW
W是大写,负载情况不清楚。
dBW=10log10P 1W Nhomakorabea 5、通道平衡
是指左右声道增益的差别一般以左、右通道输出电平之间最大值来 表示。
6、动态范围
信号最强的部分与最微弱部分之间的电平差。
二．FM参数说明
1、可用灵敏度（usable sensitivity)
接收调谐器的一项技术指标。指为使音频信号的信号/噪声（S/N）
。 比达到30dB而在天线两端所需的电压
一. 音频参数概念
1、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。
3、信噪比
指音源产生最大不失真声音信号强度与同时发出噪音强度之间的比率 称为信号噪声比，简称信噪比（Signal/Noise），通常以S/N表示，单位为 分贝（dB）。对于播放器来说，该值当然越小越好。
4、通道分离度
指双声道之间互不相干扰信号的能力、程度。通常用一条通道内的信 号电平与泄漏到另一通道中去的电平之差表示。
33信噪比信噪比指音源产生最大不失真声音信号强度与同时发出噪音强度之间的比率指音源产生最大不失真声音信号强度与同时发出噪音强度之间的比率称为信号噪声比简称信噪比称为信号噪声比简称信噪比signalnoisesignalnoise通常以通常以snsn表示单位为表示单位为分贝分贝dbdb
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-。
dac测试方法
dac测试方法DAC（Digital-to-Analog Converter，数字到模拟转换器）是一种电子设备，将数字信号转换为模拟信号。
在音频设备中，DAC负责将数字音频信号转换为模拟音频信号，以便于扬声器或耳机等模拟设备进行播放。
进行DAC测试的目的是确保DAC的性能和准确度，以保证音频信号的高质量转换。
以下是一些常见的DAC测试方法：1. 信噪比测试：信噪比是衡量DAC性能的重要指标之一。
测试过程中，将输入一个固定的音频信号，然后测量输出信号中的噪声水平。
较高的信噪比表示DAC能够更准确地转换数字信号并减少噪声。
2. 频率响应测试：频率响应测量评估DAC在不同频率下的输出准确度。
测试中，输入一系列频率的音频信号，然后测量输出的幅度和相位。
通过比较输入和输出信号之间的差异，可以确定DAC在不同频率下的性能。
3. 线性度测试：线性度测试用于评估DAC的线性转换能力。
在测试中，输入一个连续的音频信号，然后测量输出信号的失真水平。
较低的失真表示DAC能够更准确地转换输入信号。
4. 动态范围测试：动态范围测试用于衡量DAC的动态范围，即DAC 能够处理的最大和最小信号的幅度差异。
测试中，输入一个具有不同幅度的音频信号，然后测量输出信号的幅度范围。
较大的动态范围表示DAC能够处理更广泛的信号幅度。
5. 抖动测试：抖动是指由于时钟不稳定性而引起的时序误差。
抖动测试用于评估DAC的抖动性能。
测试中，输入一个稳定的音频信号，并测量输出信号的时序误差。
较低的抖动表示DAC能够更准确地转换输入信号。
以上是一些常见的DAC测试方法，通过对DAC进行全面的测试和评估，可以确保音频设备提供高质量的模拟音频输出。
这些测试方法可以帮助制造商和工程师在开发和生产过程中，确保DAC的性能和准确度达到预期水平。
蓝牙测试仪-蓝牙音频测试
− Initial Carrier Freq. Tolerance df0
(Pavg, Pmax, Pmin)
− Modulation df2 (avg, min)
12
Audio Sink (SNK) 接收者
选Audio SNK
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14
立体声测试界面 CH1:左声道。CH1:右声道
高级音频分布协议
测试内容
Audio Test
− Audio frequency (20 Hz ~ 20
kHz)− Audio 源自evel (-70 ~ 3 dBm0)
− Distortion
− SINAD
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RF Test:
− Output Power (Pavg, Pmax,
Pmin)
Audio Source (SRC) 发送者
选择Audio Source
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17
Pairable （DUT主，TC-3000C从）
总结
TESCOM可提供全套蓝牙音频测试方案
测试的参数可准确反映蓝牙音频质量， 如：检测扬声器内部的非线性等 适合研发和生产使用
北京联华行
2024/11/5
Audio Measureme nt results
DUT TX Power measured on three channels
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7
Userdefinable tone level
DUT RX level is controllable to test DUT RX performance
TC-3000C——蓝牙音频测试
• 单声道（Mono） • 立体声（Stereo）双声道 • 普通模式 (Normal Mode) 音频测试+部分RF • 测试模式（Test Mode）射频测试 • DUT：蓝牙耳机、蓝牙音箱等 • DUT尺寸、结构？ • 麦克风（Mic）、扬声器（Speaker）位置？ • 寄来实物测试实验 • 设计夹具 • 系统报价 • 北京联华行
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。
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。
美格信声学测试使用说明教程
美格信声学测试使用说明教程使用说明教程：美格信声学测试步骤1：安装美格信声学测试软件步骤2：连接音频设备在进行测试前，您需要将您的音频设备连接到计算机上。
您可以使用USB线连接您的音频接口或将您的设备与计算机配对，并确保设备已正确连接。
步骤3：打开测试界面一旦您将音频设备连接到计算机上，您可以打开美格信声学测试软件，并进入测试界面。
您可以在软件的菜单中找到“测试”选项，并单击进入测试界面。
步骤4：选择测试模式在测试界面中，您可以看到多种测试模式可供选择。
其中包括频率响应、失真度、信噪比等。
根据您的需求，选择适当的测试模式。
步骤5：设置测试参数在选择测试模式后，您需要设置一些测试参数。
这些参数将影响测试结果的准确性。
一般来说，您需要设置采样率、采样精度和测试范围。
确保您已选择合适的数值，并依据您的音频设备的规格进行设置。
步骤6：进行测试一旦您完成了上述设置，您可以开始进行测试。
根据您选择的测试模式，您可能需要播放一段特定的音频以进行测试。
确保音频的音量适中，并在测试期间保持稳定。
步骤7：分析测试结果在测试完成后，美格信声学测试软件将会显示测试结果。
您可以查看频谱分析图、信号失真图和信噪比图等结果。
根据这些结果，您可以对您的音频设备进行评估，并了解其质量和性能。
步骤8：保存测试结果最后，您可以选择将测试结果保存为文件。
这样，您就可以稍后进行比较或进一步分析。
通过点击菜单中的“保存”选项，您可以选择保存为特定的文件格式和位置。
结论：使用美格信声学测试进行音频设备测试非常简单。
通过按照上述步骤，您可以迅速而准确地评估您的音频设备的质量和性能。
这将帮助您做出更好的决策，并为您的音频制作工作提供有力的支持。
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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。