Intermittent low volume and poor frequency response

I have a 10 month old 30 Gig video IPOD that drops the volume level and loses all its low frequency response at times and then will recover after a while. This happens on a song or songs that I know are properly leveled and that work sometimes and not at other times. The issue seems to be much more prevalent when IPOD is cold or cool. After it has been operating for an hour or so the problem seems to go away. It is very common when first turned on. I am getting ready to return it while it is still under warranty but thought maybe someone has an easy fix.

Returned for warranty repair.

Similar Messages

Where can I find the amplitude and phase frequency response when the AC coupling is set on the 4472

My friends,
I need to know the amplitude atenuation but most important, the phase distortion introduced on the low frequency components of the signals when the AC coupling is setted (i.e. when the high pass filter is connected) on the 4472 and 4472B DAQBoard.
I can construct the amplitude frequency response by generating and aquiring a sine waveform of a knew amplitude. But I cannot construct the phase distortion introduced by the circuitry.
However, I assume that this crucial information should be available in the DAQ Manual or in the website of NI, but untill now I haven´t found it.
Thankning in advance,
crimolvic from Chile

crimolvic,
Here are the Specifications and Datasheet for the 4472. They indicate a phase non-linearity of less than 0.5 degrees across all frequencies.
For information on how this varies with frequency, see the attached
spreadsheet. This response was the result of testing on a single
4472. Although this is classified as a "typical" response, it is
not gauranteed.
Have a great, day!
Travis
Attachments:
4472 Phase Linearity.xls ‏21 KB

Game is running with low resolution and poor performance

Hi, I have purchased dv4 entertainment laptop 2 years back. Recently, I installed Ford Collections - Car game in my laptop, but it is not running properlly which means that the resolution is little poor and it is very poor and getting slow when playing multiplayed in the laptop. Also, I am unable to install the Adventures game in my machine, getting DirectX error. I am not sure this issue is because of the in-adequate configration of Graphics card Please guid me to reslove this issue.

Compressed using compressor, burned DVD with I-DVD.
Compressed to what codec using Compressor?
iDVD uses its own internal encoder -it has probably gone through two compression cycles as a result, which will hammer the quality.
You should feed iDVD with a file from FCP using this method instead:
File > Export > QuickTime Movie > Current Settings. You can choose Self Contained or Reference.
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Muvo TX is suffering from low volume and then volume spi

Can anyone tell me if volume spikes are a common problem? As my headset jack rotates around the outlet (both are round and therefore capable of moving 360 degrees) during the course of normal wear the volume goes from being extremely soft to headache causing loud. Does anyone know what is causing the problem and how to fix it's
I've already tried different headsets to make sure they were not the problem.

This is not normal behavior. I have a 256mb TX that doesn't do this. Do you have the most recent firmware installed(.0.05),not sure if this would help or not. Also, can you test the .mp3's on your computer to make sure that the volume is about the same for all files?

Beats in software tracks seems really low volume and abrupt

If i use the keyboard to play beats, they seems really electronic - meaning abrupt- i would like them to gradual - like it should sound like they start, hit a peak and then fade, not simply "boom", is there any way i can do this? also, the volume of the software track is not that great, max volume is causin clipping, Manu

If i use the keyboard to play beats, they seems really electronic - meaning abrupt- i would like them to gradual - like it should sound like they start, hit a peak and then fade, not simply "boom", is there any way i can do this?
What drum set are you using? The drum sets consist of samples, they are triggered with every note, no matter how long or short it is, you can only change the velocity ("volume"). However, some drum sets have pretty natural sounding sounds, maybe you're using an electronic or hip hop kit.
also, the volume of the software track is not that great, max volume is causin clipping
Well, then turn the track volume down. It's not meant to be cranked up all the way all the time. The opposite would be really bad - if you couldn't hit the ceiling no matter how you turn up the volume.

Frequency response: "very poor" on SB audigy

just encountered this very strange thing. Right Mark Audio Analyzer 5.5 reports and diagrams a "very poor" frequency response on my SB0570 Audigy at 44 Khz and 6 bits (not only) loop-back test.
http://img86.imageshack.us/img86/6409/rmbenchmark3pv.jpg
is this normal?

walkietv wrote:
ahaa! thanks. the problem is the guide is pretty old and the mixer and EAX look different. Could you tell me please exactly what should i change. I mean what to put in REC, Source in the mixer. I checked the EAX console-only time scalling was acti've. Any other setting in EAX? thank you!
- all options on all EAX panels tabs needs to be set 'OFF' (i.e. not enabled).
- if you're running the loop back test (line-out --> line-in), the Rec -source shall be set to AnalogMix() (or if you see the Line-In mentioned there as separate source then select it).
- you can set the monitor while recording by pressing the '+' sign above the Line-In slider (Surround Mixer) --> disable this option (i.e. no monitoring while measuring)
Other settings:
- windows audio performance options --> set acceleration and SRC to 00%
- on Surround Mixer, set master volume 00%, Source: Wave 00%, Line-In 50% and rec source 50% (adjust the signal level to ~ -dB by moving the Line-In or/and rec sliders)
jutapa

Hi. I Have a Pavilion dv9000 which has always had very low volume - really needs to be increased?

Hi. I Have a Pavilion dv9000 with Vista Ultimate 64. It has always had very low volume and I find I generally have to have the volume up into the 70's 80's as well as the onscreen media player slider almost full - which then gets all distorted - and then have to turn it down again when not using the player as the computer error beeps come screaming out., Ive checked the settings on speakers to make sure all are the same & balanced and also have updated any codecs as needed. Is there a sound lever setting elsewhere?
cheers

Thanks for the reply and peeling my post off Mike, sorry I didn't start my own initially.
Well @ 1333Mhz I was using the rated timings 9-9-9-24 (as per Corsair) but it wasn't very stable as I said.
I have since changed the FSB/RAM ratio to get 1066Mhz and it seemed much happier. (@1.5v) It would be nice to run the RAM at it's rated setting but my OC skills aren't that extensive. I imagine it would need alot of tweeking? Or could I have just relaxed the timings a bit?
Would removing the Switch Cards make any difference?
Thanks in advance.

Frequency response function modal analysis

After reviewing the signal analysis functions in DIAdem I have realized them to be a bit limited for modal analysis. I have a couple hammer impact tests that I need to process a frequency response function for, and since this is brand new to me I'm not seeing anything in the embedded function list that is going to help me. I was wondering if anyone out there has a couple of pointers on generating a FRF plot for modal hammer impact tests. I did notice that the ChnFFT2 command allows me to generate a transfer function, coherance, and FFT Cross Spectrum channels for analysis. Though I might be confused and this may be everything I need. My FFT2 settings are below.
[code]
FFTIndexChn      = 0
FFTIntervUser    = "NumberStartOverl"
FFTIntervPara(1) = 1
FFTIntervPara(2) = 2500
FFTIntervPara(3) = 1
FFTIntervOverl   = 0
FFTNoV           = 0
FFTWndFct        = "Rectangle"
FFTWndPara       = 10
FFTWndChn        = "[1]/Time axis"
FFTWndCorrectTyp = "No"
FFTAverageType   = "No"
FFTAmplFirst     = "Amplitude"
FFTAmpl          = 1
FFTAmplType      = "PSD"
FFTCrossSpectr   = 1
FFTCoherence     = 1
FFTTransFctType = "Spectrum H0"
FFTCrossPhase    = 0
FFTTransPhase    = 0
Call ChnFFT2("[1]/Time axis","'[1]/H_1' - '[1]/H_4'","'[1]/A_1' - '[1]/A_4'") '... XW,ChnNoStr1,ChnNoStr2
[/code]

Standard modal analysis has something denoted as FRF. I have a labview application note "The Fundamentals of FFT-Based Signal Analysis..."
Frequency Response Function
The frequency response function (FRF) gives the gain and phase versus frequency of a network and is typically computed as
where A is the stimulus signal and B is the response signal.
The frequency response function is in two-sided complex form. To convert to the frequency response gain (magnitude) and the frequency response phase, use the Rectangular-To-Polar conversion function. To convert to single-sided form, simply discard the second half of the array.
You may want to take several frequency response function readings and then average them. To do so, average the cross power spectrum, SAB(f), by summing it in the complex form then dividing by the number of averages, before converting it to magnitude and phase, and so forth. The power spectrum, SAA(f), is already in real form and is averaged normally.
Refer to the Frequency Response and Network Analysis topic in the LabVIEW Help (linked below) for the most updated information about the frequency response function.
http://zone.ni.com/devzone/cda/tut/p/id/4278
So the options for FFT2 are
No
DIAdem does not calculate a transfer frequency response.
Spectrum H0
DIAdem calculates the transfer frequency response by dividing the FFT of the output signal (A) by the input signal (E): FFT(A)/FFT(E). DIAdem averages the amplitudes of the individual transfer functions.
Spectrum H1
DIAdem specifies the cross spectrum and the auto spectrum for each signal pair. DIAdem calculates the transfer frequency response by dividing the averaged spectra: Middle(cross(A,E))/middle(auto(E)). DIAdem does not average phases, because phases can delete each other.
Spectrum H2
DIAdem specifies the cross spectrum and the auto spectrum for each signal pair. DIAdem calculates the transfer frequency response by dividing the averaged spectra: Middle(auto(A))/middle(cross(E,A))
If you assign the values Spectrum H1 or Spectrum H2 to the variable FFTTransFctType, DIAdem averages and divides the cross spectra and the auto spectra and calculates the amplitudes last.
Which state auto spectrum when FRF is power spectrum.

Xperia Z1 mic issue (low volume) when using WeChat

Was searching this forum for a solution, but to no avail. Using Wechat on Xperia Z1 with Android 5 is impossible because mic records at very low volume and other parties strugle to hear me. Microphone suppresion and voice enhancement are turned off, equalizer set to normal. PC companion shows most current version of software. Dear users and/or Sony, could you suggest for any solution?

Please consider this before buying Sony Xperia phones: Chat TranscriptPlease wait while we find an agent to assist you...You have now been connected to an agent.Emir: Hi Client Name! Welcome to Sony Xperia Support, my name is Emir.
I'm currently reading your message. In the meantime, I would appreciate if you kindly could leave your first and last name, your e-mail address and mobile phone number.Client Name: Client Name, [email protected], +37067530xxxEmir: How may I help you?Client Name: Hello, I have posted a topic about an software issue in Xperia Z1: http://talk.sonymobile.com/t5/Xperia-Z1-Z1s/Xperia-Z1-mic-issue-low-volume-when-using-WeChat/m-p/1020748/highlight/true#M44196Client Name: it was ignored by staff, so I posted another topic about how Sony ignores issues of flagship phone ownerClient Name: this topic was moved out of forum to this topic to be burriedClient Name: can I speak with someone in charge?Emir: No, there's no such contact I can refer you to,.Emir: I'm sorry to hear that your thread has not been replied to.Client Name: What do you mean? You're independant worker not accountable to anyone?Client Name: its been replied with default answer that all posters with similar problem receivedEmir: I'm a technical support agent for the chat, I don't have any information about the Forum team.Client Name: we are specifically ignored until Z1 goes broke and out of your problemsClient Name: I need someone in charge from your department who could personally overlook this issue that many users haveClient Name: its not just me, mic problems are apparent for many usersEmir: This issue concerns a third party app, and you have already been in contact with me regarding this issue. We can't guarantee that third party apps are compatible with your device since we do not develop them. I recommend that you contac the app developer for more information.Client Name: I dont understand why phone hacker can solve this software issue by themselves, but Sony engineers choose to ignore usClient Name: no, this issue also goes to many other apps, including native app TrackIDClient Name: its the problem in new Android releaseEmir: You may try a software repair to see if the issue persists in TrackID like the agent on the forum suggested - if the problem persists and you suspect there's something wrong with your device you may have it sent in to an authorized service center so that a technician may examine it.,Client Name: These steps were performed by other users with no luckClient Name: I wont repeat this to come to the sameClient Name: I ask engineers to look into solution developed by XDA team and implement for us normal usersEmir: Those are the troubleshooting steps I can refer you to in this case.Client Name: They dont work, please find another solution or ask for person who can do it, if you cantEmir: I have already provided troubleshooting steps, you may try that. If a software repair does not help then I can only refer to service.Client Name: I already told that these steps didnt work for other users, who have same problem. So you admit that you wont help your client?Emir: I can't suggest a possible solution other than what I've already suggested unfortunately.Client Name: So I have no choice, but to share our chat in all possible internet sources for other existing or would be Sony users to consider how they will be treated upon problems arise.Client Name: Thank you for your support, I hope you will be promoted for high quality service

Very low volume, 845PE-MAX2BFIR

Hi, I have just bought a MSI 845PE-MAX2 BFIR and I have connect 2 speakers to the green "line out/front speaker out", it only sounds a very low whisper sound from one of the speaker even then the volume is turn to maximum, why?
When I was trying to changes some settings to use Xear 3D 4 channel then I needed to changes the connection of the cable to the S-bracket, and at this place I can hear the very low whisper sound from both left and right speaker, but still the volume is to low, what can be wrong?
On the connection it says "line out/front speaker out", how to be sure it is the front speaker that I use? In the setup I have choice I have 2 table speakers to my computer but this doesnt help.
Hope anyone can help me.
Thanks
/Peter

Quote
Originally posted by pekn
Hi, I have just bought a MSI 845PE-MAX2 BFIR and I have connect 2 speakers to the green "line out/front speaker out", it only sounds a very low whisper sound from one of the speaker even then the volume is turn to maximum, why?
When I was trying to changes some settings to use Xear 3D 4 channel then I needed to changes the connection of the cable to the S-bracket, and at this place I can hear the very low whisper sound from both left and right speaker, but still the volume is to low, what can be wrong?
On the connection it says "line out/front speaker out", how to be sure it is the front speaker that I use? In the setup I have choice I have 2 table speakers to my computer but this doesnt help.
Hope anyone can help me.
I've got a 845PEmax2 - SR with almost the same complaint. With attached s-bracket a lot of noise and low volume and without s-bracket attached normal sound and all the connections on the back-panel.
I still do'nt know what is causing this trouble.
Is there soneone from MSI who could help us with this (hardware) problem?
herman

3rd Generation Low volume ■Frequency response: 20Hz to 20,000Hz

Hi - My slightly used EBAY Purchased IPod Nano 3rd Generation plays some songs to my cars volume capacity; and most songs in very low volume {no matter what I do in settings/EQ, ect}.
Could this have something to do with how I purchased the song in ITunes? What exactly is bitrate ACC on the sync menu and how do I know to choose that option.
Side Note: I have a Harmon Kardon system in a Landrover LR3~ which means this 43 year old mom of 2 should be able to feel the beat when driving. The automobile radio/CD/XM can perform, why can't my IPOD?
Humbly Needing to Groove While a Cruise,
Donna

Having been an audiophile for decades I can tell you specs lie. Cheap audio equipment often has specs that ON PAPER best units costing literally 100 times as much. It's best to read trusted sources for reviews and use your own ears. I learned about the "arms race" in specsmanship back in the 70's when I was in and out of high end audio salons for years, replacing components in my home system. I finally got what I really like when I got, among other things, a Mark Levinson amp. NOTHING beats listening and since our ears have short term memory, the best way to know which of several pieces of equipment are best for you is to do a carefully controlled A/B test. A/B tests CAN be biased, but done fairly (no salesman shenigans) they are the best way to know which item to buy (presuming your pocketbook matches the quality of your hearing)
Message was edited by: David.
Message was edited by: David.

Frequency response at low frequency

I'm working on a bandpass filter, and I'd like to get the frequency response showing that the frequencies outside the lower and higher cutoff frequencies are being cut off. However, the correct plot is shown only when my cutoff frequencies are high (roughly from 1000-8000 hertz). When I use low cutoff frequences(roughly 4-5 hertz), the plot is incorrect. So how can I get the frequency response to my low cutoff frequencies? Thanks.
P.S. In the code, some parts are irrelevent. In the front panel, the only relevent part is the frequency response plot at the lower right corner, and the specs above it; in the block diagram, only the upper half(with IIR and FRF) is relevent. Thanks.
Attachments:
BME_Pressure_Sensor_V1.00.vi ‏591 KB

Hi Manson
There is a bug in your diagram since you connected the number of samples where you should have connected the sampling frequency.
The sampling frequency is related to the pace at which you take the measurement.
Usually, Fs = 1 / dT
where Fs is the sampling frequency and dT is the time interval.
It should work better.
In any case, to have a better resolution in the low frequency range of your spectrum computations, you have to increase the number of points of your data because there exist the following relationship between dF (space between 2 points in you spectrum), dT, and N (number of data points) :
dF = 1 / (2 x dT x N)
Doc-Doc
Doc-Doc
http://www.machinevision.ch
http://visionindustrielle.ch
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Frequency response - sound and vibration

Hi,
I need to find the frequency response of the DUT. I am using the NI example from sound and vibration toolkit to do
so (LabVIEW 8.5\examples\Sound and Vibration\Getting Started\SVXMPL_Getting Started with Swept Sine (DAQmx).vi) now my problem is
to tabulate the phase difference between the stimulus and response when i do it i get constant values. Even though i dont give any response
signal to the channel i get the wave same as stimulus, is it suppose to be like that !!!!!!!!!! I will attach my vi so that it gives the idea where
i am measuring the phase difference pls check it and help me with this.....thank you in advance.
Attachments:
DAQ Freq resp (req)_trial.vi ‏121 KB

What is the DAQ card you are using for executing this Frequency Response?
This DAQ error is related to the output frequency of your signal higher than the possible output rate of the card. Basically, you are trying to update at a rate higher than the capability of the card.
For doing Frequency response, generally you need to have a NI DSA card (446x/447x/449x) with one Analog output and two analog inputs (minimum). Generate the swept sine signal from the Analog output channel, give this signal as input to your DUT and also to one of the Analog input channel. This input to your Analog input channel will act as your Reference signal. Then the response signal from your DUT, connect it to another Analog input channel. You would get a very good response results.
The reason why you need to have a DSA board is that, for doing Frequency response, we need to acquire both high frequency and low frequency components without much loss. This is possible only if your DAQ board has a higher dB value (in the range of >110dB) which is present only in DSA boards.
I have completed a Frequency Response Analysis just a week back with the same example programs. So there wont be any problem with that vi.
Regards,
Sundar Ganesh

Calculate frequency response using FFT and inverse FFT

Hi,
Attached is the program using FFT and inverse FFT to filter a time domain signal. The frequency response of the LPF can be obtained by using the chirp signal from 0 to 5kHz. However, I don't know why the signal obtained from a sine wave input is so strange. The amplitude is wrong and has a envelope outside. Please help to point out what's wrong with that.
Bill
Attachments:
fft filter.vi ‏87 KB

If you check the help text for sine wave.vi you'll see that it generates the sine wave based on the following formula:
yi = a*sin(phase[i])
for i = 0, 1, 2, …, n – 1 and where
a is amplitude,
phase[i] = initial_phase + f*360*i
This means that when you input a=1, f=0,1 and initial_phase=0 you will get a sine wave that is based on samples at every n*36 degrees; i.e. at 0, 36, 72 etc...due to this sample rate you never see the full amplitude (+/- 90 degrees), the wave is clipped at the top. If you input an initial phase of 64 degrees you will get the full amplitude, but the wave is still deformed due to digitalization...
The lower the frequency you put in, the closer the digitalized representation will be to the true sine.
Use the Waveform Generator VIs from the analyze palette if you want to have more control over the wave generation (sample rates etc.). (Not available if you have the base package.)
MTO

When I have headphones plugged into my iphone it says warning'' high volume'' and becomes first a red and then plop a number of red pops but it will not be so for someone else I know. My sound will be much lower than anyone else in the music, and when I t

When I have headphones plugged into my iphone it says warning'' high volume'' and becomes first a red and then plop a number of red pops but it will not be so for someone else I know. My sound will be much lower than anyone else in the music, and when I talk by phone headset

Hi! I have the same problem when I use my headphones .
iPhone 4s England, iOS 6.1.2

Intermittent low volume and poor frequency response

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