After starting hearing sound becomes distorted (sample rate problem?)

Similar Questions

• CANOpen - sample rate problem

  Hello

  I use the CANOpen toolkit to communicate with a sensor that can be set at a sampling frequency of 1 kHz. Before I used the standard frame API and there I received my PDO objects to each MS because I want to make programming easier I would use CANOpen. But I can't set the sample rate of 1ms. When you use 2ms data is not transmitted with the correct timestamp interval.

  I have attached the message error code when you use 1ms as sampling frequency and the part of the diagram block with the initialization of the PDO.

  Maybe someone can give me a hint.

  Best regards

  --
  Joachim

  It is good, your sensor begins to send pictures as soon as you start communication.

  You might remove the remote setting function and then it would behave like your old application framework API, except for one detail.

  Your old approx. api framework its mult read read all available images.

  The CANopen Read is only a single point read. Thus, he read only a single image. This means that you need to run the loop in ms rate, that is really risky with windows.

  You can enable buffering, but it can lead to errors of overflow, or turn off the buffering is now and read the most recent value.

  See the example updated the.

  DirkW

• With the NI 9205 module Max sampling rate - problems

  Dear friends,

  I develop a project of lv, which makes and control system of engine dyno. The material is CRio-9022 with other cards and also 9205 for AI. There is an encoder for angle attached to the motor shaft with 3600 chatted by Tower as well as an index to indicate the end of a revolution. the output of the encoder is measured by card 9411. The speed of the motor is 1500 rpm. I measure pressure data and couple when I receive a 'tick' of the wheel. This means my sampling rate for pressure and torque each is 90KO/s.

  but I was not successful to lead it. The program is great and I can show them, but I believe that there is a problem in the choice of material for the task. With the data of pressure and torque of the 9205, I also measure other channels for the controller output mass flow and temperatures. So in all I use 8 channels of the 32 available. But only the pressure and torque are acquired at the wheel-driven sampling rate. the rest are acquired about 5 times per second.

  Since the 9025 is a multiplexing ADC, 250K sampling frequency is divided by the number of channels accessed = 250 K/8 = 31 K samples/channel. With this in mind, I decided to acquire data of pressure and torque with each beat 3rd rotary encoder, essentially on 30K samples/s sampling. However, I see a large amount of noise.

  So I decide to average more than 1 second cycles (so the engine runs at about 25 cycles/sec, I averaged over this issue). The resulting pressure and torque graphics do not match with those measured by an oscilloscope in terms of amplitude but the frequency and shape is correct.

  I noticed an interesting feature in the charts. When I pass interpolation between the points, I see several curves made by points instead of a continuous locus of points. Accordingly, I find that the acquisition is slower than necessary, and so there are less number of points sampled as required. These points are not synchronized 25 cycles I have on average and therefore the separate "curves". It is because of the possibility that some points receive a higher number of 'contributions' several times (when you add), the neighbouring points.

  so I conculde that the 9205 is not fast enough to do the job. also noise, perhaps due to crosstalk or gosting when the mux changes channels. the impdences output pressure and the couple are of the order of 10 K ohms.

  the Labview code outline: well, there is a vi FPGA, which takes the rotary encoder ticks and sends a signal to the case of each 3rd tick. The signal contains a 16-bit integer, indicating the number of ticks. This signal is sent to a 1 element FIFO. This fifo is read in a parallel while loop, where it remains awaiting a new element. The while loop bed fifo, where data are available, takes a measure of pressure channel. A node memory of the method is called to provide data according to contained in the index number equal to the number of ticks to signal fifo. Then he adds the current pressure reading to the reading of the memory and stores the sum in the same memory location.  Thus, an array of elements of 1200 is formed, where each elemnt is a sum of the values taken of more than 25 cycles. This memory is transferd to a dma fifo and reading side host. is done similarly to involved couple. host-side the fifo is read and divided by 25 to get the average. This average is displayed on a waveform graph.

  Please check the attached file to get an idea of the problem. Sorry for the long post.

  Please suggest if you understand the problem and suggesstions or solutions.

  
  

• can someone tell me why the sound becomes distorted on my external speakers when I plug the charger on my netbook?

  I have a set of logitech speakers and a subwoofer that I use on my netbook, but when I plug the charger to recharge the battery, the sound distorts seriously which means I can listen only as long as the battery, I sent l; ogitech but they said it was a problem on my computer. Anyone have any ideas why?

  Hi gaz2558,

  ·         Have you made changes on the computer before this problem?

  Follow the steps in the article.

  http://support.Microsoft.com/kb/813498

  Also update the latest audio drivers.

  http://support.Microsoft.com/kb/166774

  Reference: http://support.microsoft.com/kb/307918/en-gb

• 9237 & 9205 with 9178 chassis sampling rate problem

  I have two 9237 and installed 9205 on 9178 chassis. In order to acquire data at the same time, all the 9237 and 9205 use the 9178 chassis onboardclock. But when the sampling frequency is 20000 Hz, 20000 points just takes less than a second. When the sampling frequency is 10000Hz, get 10000 points takes 1 second.   Why does this happen?

  A 9237 use a channel, another 9237 use 4 channels, the 9205 use 3 channels (differential).  And manual brand 9178 slightly confused me, the frequency of onboardclock MHz 80?

  Looking forward to your help. Thank you.

  By default the 9237 derives its clock dividing down from his own time of 12.8 MHz (see specifications) base.

  12.8 MHz / 256 / N

  Where N is a whole 1 to 31.

  10 kHz is achievable (12.8 MHz / 256 / 5), but it's not 20 kHz (12.8 MHz / 256 / 2.5).

  The driver will round up to the next frequency, which is be of 25 kHz (12.8 MHz / 256 / 2).  25 kHz, we would expect 20 k samples to be acquired in 0.8 seconds.

  Best regards

• Sampling rate of change

  I'm trying to use CC2015 (currently use CC2014) but I am struggling to change the sampling frequency.  It is easy to change w/2014.  It is fixed at 48 k in CC2015 and I would like to change from 44.1.  There is no option for this in the drop down menu (although my card supports the lower rate).  I'm using a saffire pro 24 dsp.  In fact, I went in the control panel of this sound card to see if I can adjust here (even if - ever due in 2014), but it does not give me an option.

  First of all, you're on a Mac or a PC? In the hardware Audio hearing set up page what should I have selected in the class of device? What sampling frequency is the watch as shown in the article of the Saffire MixControl State of the device? You should be able to change the sample here, but maybe not while the hearing is open.

  Also on the page of audio of the hearing you have the "strength material attempt to document sampling frequency" box checked or unchecked? Try tick or untick this box.

  Unfortunately, I think that CC 2014 has tried to protect the user against rate changes of sample size which means that sometimes hearing automatically made a sampling rate Conversion if the material match those of the hearing. It wasn't really a good idea, even if it is easier for the novice user. So now you may have to manually adjust both.

• After that start monitor/screen becomes black with visible vertical lines

  After that start monitor/screen becomes black with visible vertical lines. Move the mouse shows an activity going on behind the disorder. My son says video card?

  Hi Melissa,

  ·         Since when are you facing this problem?

  ·         Have you checked the issue by connecting a different monitor?

  Step 1: It seems to be a hardware issue to me. Check if the cables are well connected.

  Step 2: Connect another monitor to the computer and see if the problem persists.

• Specific sampling rate?

  So maybe this is a stupid question, but I need to know because I train for a specific sound. Is there a way [to logic] to shoot/change of a certain frequency sampling rates. I can imitate the sound I'm looking for with a low pass filter, reverb and a distortion. But I don't want to 'emulate', this sound, I want to create. Then I can put my own effects and play with it like I want to. If I have to use a bunch of effects to make it sound like I want that also the addition of said effects remove the sound and sound horrible. as to where pulling the sampling frequency of the high frequency and no downs will make me THE noise that I need and always allow to add nice effects to make MY sound instead of someone else. I hope you know what I mean. Let me to you specific real once more. I want to pull or carry a certain frequency sampling rates for a sound under water. I don't want to use filters to make the sound. So can you please help me. I invited everyone locally on how to do it and nothing works. Also if this is not possible in the logic of tell me if there are third party plug ins or maybe even a different DAW that could do like komplete Kontrol or audacity.

  Effect under water

  See if this thread is helpful at all...

• Writing data to extend the acquisition of data for the sampling rate high file

  These are the tasks that I have to do to take noise measurements:

  (1) take continuous data to USB 6281 Office, in a sample of 500 k (50 k samples at a time) rate.

  (2) save data continuously for 3 to 6 hours in any file (any format is OK but I need to save in a series of files rather than the single file). I want to start writing again file after every 2 min.

  I enclose my VI and pictures of my setup of the task. I can measure and write data to the file continuously for 15 minutes. After that, I see these errors:

  (1) acquisition of equipment can't keep up with the software (something like that, also with a proposal to increase the size of the buffer...)

  (2) memory is full.

  Please help make my VI effective and correct. I suggest to remove him "write in the action file" loop of consumption because it takes a long time to open and close a file in a loop. You can suggest me to open the file outside the loop and write inside the loop. But I want to save my data in the new file, after every 2 min or some samples. If this can be done efficiently without using Scripture in the measurement file then please let me know.

  Thank you in advance.

  This example here is for a single file and a channel, you should be able to loop over that automatically. The background commentary should be the name of the channel, no group namede the name of the channel in the control.

  

• . VI filtering IIR and response: response of Butterworth filter size depends on sampling rate - why?

  Hi people,

  I'm not an expert in the design of the filter, only a person in applying them, so please can someone help me with an explanation?

  I need to filter signals very infrequent using a buttherwoth filter 2. or 3. order of the bandpass 0.1 to 10 Hz.

  Very relevant amplitudes are BELOW 1 Hz, often less than 0.5 Hz, but there is as well the amplitudes beyond 5 Hz to observe.

  It's fixed and prescribed for the application.

  However, the sampling rate of the measuring system is not prescribed. It may be between say between 30 and 2000 Hz. Depends on the question of whether the same set of data is used for analysis of the higher up to 1000 Hz frequencies on the same measure or this is not done by the user and he chooses a lower sampling rate to reduce the size of files, especially when measuring for longer periods of several weeks.

  To compare the response amplitude of 2nd and 3rd order filter, I used the example of IIR filtering .vi and response:

  I was very surprised when I found that the response of greatness is considerably influenced by the SAMPLING RATE I say the signal generator in this example vi.

  Can you please tell me why - and especially why the filter of order 3 will be worse for the parts of low frequency below 1 Hz signal. Told me of people experienced with filters that the 3rd oder will less distort the amplitudes which does nothing for my the frequencies below 1 Hz.

  In the attached png you see 4 screenshots for 2 or 3 command and sampling rate of 300 or 1000 Hz to show you the answers of variable magnitude without opening labview.

  THANK YOU very much for your ANSWERS!

  Chris

  Hello Cameron and thanks for my lenses of compensation.

  I can now proudly present the solution of my problem.

  It seems to be purely a problem of the visualistion information filters through the cluster of the scale.

  After looking in the front panel of the IIR, I suddenly noticed that the "df" of the pole size is changing with the Fs of the input signal.

  For a Fs to 30 Hz, the "df" is 0.03 Hz so you see the curve of the filter with more points, see png.

  For a Fs 300 Hz "df" is 0.3 Hz, so the curve is larger with only 3 points between 0 and 1 Hz.

  For a 1 kHz Fs the df is 0,976 Hz, so there is no point in the graph between 0 and 1 Hz.

  It's strange that for constant Fs, df of this cluster NOT reduced with the increase in the number of samples, as it does in an FFT.

  However, I hope now the filter used now for the curves obtained with the proposed Lynn way and the response of greatness from the filter information fit together.

  Thank you for your support.

  Merry Christmas and a happy new year to all.

  Chris

• The PXI - 6259 sampling rate M SERIES DAQ

  Good afternoon friends,

  I'm reading my series M manual trying to find AO maximum sampling rate I can run my PXI-6259 to.   I need to generate four-channel (A0:3) grid output wave high fidelity.   Higher sampling frequencies enable higher frequency in tone generation and a better representation of the sounds of broadband as white noise.   How fast can I run this puppy, and is there a point where performance and reliability starts to suffer?

  I'll keep digging for answers!

  Thank you

  Zach

  Zach Hey!

  If you look on page 3 of the 625 x card product, the analog output with 4 channels update is 1.25 MECH. / s.  Rock on!

• When I play games in Windows Vista, the game freezes and the sound is distorted. I have

  I have problems playing my games in windows vista 7 the sound is distorted and the game freezes. It becomes very slow.  32-bit computer laptop Dell Inspiron

  Hello

  You can check if the third-party application is causing games freeze. To identify which application can be the question originally configures your computer to clean the start State and follow the troubleshooting steps in the article. To configure your computer to clean boot follow the link below.

  How to troubleshoot a problem by performing a clean boot in Windows Vista or in Windows 7

  http://support.Microsoft.com/kb/929135

  Note: When you are finished troubleshooting, follow step 7 article to start on normal startup.

  Hope this information is useful.

  Amrita M

  Microsoft Answers Support Engineer
  Visit our Microsoft answers feedback Forum and let us know what you think.

• Music or the sounds are distorted (Echo, off the field, sound effects)

  When listeing music using headphones/earphones sound effects are terrible it resonates, off-sites. Sometimes, so the videos of youtube listeing and listeing to music/videos stored on my system. I don't remember if I did configuration changes in the sound of the control panel. But I am powerless to return to normal.

  Here are the details of the required system.

  Model: HP Pavilion Notebook PC g6.

  BIOS: F.26

  Processor: AMD A10 - 4600 with Radeon HD graphics 4cpus ~2.3 Ghz APU

  Memory: RAM 8192 MB

  DirectX 11

  O.S: Windows 8.1 only 64-bit language

  -------------
  Sound Devices
  -------------
  Description: Pregnant / HP (IDT High Definition Audio CODEC)
  Default audio playback: Yes
  Default voice playback: no
  Hardware ID: HDAUDIO\FUNC_01 & VEN_111D & DEV_76D9 & SUBSYS_103C184A & REV_1001
  Manufacturer ID: 1
  Product ID: 100
  Type: WDM
  Driver name: stwrt64.sys
  Driver version: 6.10.6425.0000 (English)
  Driver attributes: Final retail
  Would be WHQL logo: Yes
  Date and size: 20/08/2012 11:15:20, 542208 bytes
  Other files:
  Driver provider: IDT
  HW Accel level: basic
  Cap flags: 0xF1F
  Min/Max sample rate: 100, 200000
  Mix beef static/Strm HW: 1, 0
  Static/Strm HW 3D steers: 0, 0
  HW memory: 0
  Management of the voice: No.
  Listen/Src/EAX (TM) 2.0: No, no
  Listen/Src/I3DL2 (TM): No, no
  Sensaura (TM) ZoomFX (tm): No.

  Description: Helmet communications (IDT High Definition Audio CODEC)
  Default audio playback: No.
  Default voice playback: Yes
  Hardware ID: HDAUDIO\FUNC_01 & VEN_111D & DEV_76D9 & SUBSYS_103C184A & REV_1001
  Manufacturer ID: 1
  Product ID: 100
  Type: WDM
  Driver name: stwrt64.sys
  Driver version: 6.10.6425.0000 (English)
  Driver attributes: Final retail
  Would be WHQL logo: Yes
  Date and size: 20/08/2012 11:15:20, 542208 bytes
  Other files:
  Driver provider: IDT
  HW Accel level: basic
  Cap flags: 0xF1F
  Min/Max sample rate: 100, 200000
  Mix beef static/Strm HW: 1, 0
  Static/Strm HW 3D steers: 0, 0
  HW memory: 0
  Management of the voice: No.
  Listen/Src/EAX (TM) 2.0: No, no
  Listen/Src/I3DL2 (TM): No, no
  Sensaura (TM) ZoomFX (tm): No.

  ---------------------
  Sound Capture devices
  ---------------------
  Description: Micro internal (IDT High Definition Audio CODEC)
  Capture audio by default: Yes
  Default voice Capture: Yes
  Driver name: stwrt64.sys
  Driver version: 6.10.6425.0000 (English)
  Driver attributes: Final retail
  Date and size: 20/08/2012 11:15:20, 542208 bytes
  Cap flags: 0x1
  Format flags: 0xFFFFF

  Description: Stereo Mix (IDT High Definition Audio CODEC)
  Capture audio by default: No.
  Default voice Capture: No.
  Driver name: stwrt64.sys
  Driver version: 6.10.6425.0000 (English)
  Driver attributes: Final retail
  Date and size: 20/08/2012 11:15:20, 542208 bytes
  Cap flags: 0x1
  Format flags: 0xFFFFF

  ---------------------
  Video Capture devices
  Number of aircraft: 1
  ---------------------
  FriendlyName: HP TrueVision HD
  Location: front
  SymbolicLink: \\?\usb#vid_064e&pid_e263&mi_00#6&39db09e7&0&0000#{e5323777-f976-4f5b-9b55-b94699c46e44}\global
  Manufacturer: Microsoft
  HardwareID: USB\VID_064E & PID_E263 & REV_0105 & MI_00, USB\VID_064E & PID_E263 & MI_00
  DriverDesc: USB video device
  DriverProvider: Microsoft
  DriverVersion: 6.3.9600.16384
  DriverDateEnglish: 21/06/2006 00:00:00
  DriverDateLocalized: 21/06/2006 00:00:00
  Service: usbvideo
  Category: Image
  DevNodeStatus: 180200A [DN_DRIVER_LOADED |] DN_STARTED | DN_DISABLEABLE | DN_NT_ENUMERATOR | DN_NT_DRIVER]
  ContainerId: {00000000-0000-0000-FFFF-FFFFFFFFFFFF}
  ProblemCode: No problem
  BusReportedDeviceDesc: HP TrueVision HD
  Mother: USB\VID_064E&PID_E263\HF1016-T821-SE01-VH-R01.00.05
  DriverProblemDesc: n/a
  UpperFilters: n/a
  LowerFilters: n/a
  Battery: \Driver\ksthunk,\Driver\usbvideo,\Driver\usbfilter,\Driver\ACPI,\Driver\usbccgp
  ContainerCategory: Imaging

  Please help solve this problem.

  Hello RPrasad, welcome to the Forums of HP.

  I see that you have a problem with the sound in your headphones with your laptop. I will try to help you with this.

  Have you tried another headset to see if the problem occurs from there also? It is preferable to exclude the logical hardware first.

  If the problem does not form several headphones so I suggest to start with this document: audio speakers (Windows 8) No. (the steps can be applied to any sound problem)

  If this fails to solve the problem, as your not sure of any changes made to the laptop. Your next option would be to do a system refresh. This document can help you on how to do it: Refresh Your PC to solve problems (Windows 8).

  I'll leave you with this, I would like to know how it goes for you.
  Thanks for posting on the HP Forums.

• Audition 3.0 how to disable ASIO and the default Sample Rate recording?

  Hi people,

  New here, but I hope someone can help me with a few questions, I'll have with Audition 3.

  Firstly, some background questions.

  I use hearing parallel to a broadcast audio broadcast program called SpotOn,

  This software requires that I run the sound card, a RME Madiface XT in 48 k mode, and that all the outputs that it uses are defined as WDM Windows so that the windows kernel mixer can control them.

  This means that when I use the hearing at the same time, I have to configure it to use the "Audition 3.0 Windows Audio" driver to stop him from taking control of the sound card directly and change setting which prevent SpotOn to see its output.

  The problems I encounter are that hearing itself seems to randomly change mode in the edit window ASIO driver, I suspect that this happens when I import audio data from a key which is from 44.1 to modify for use in SpotOn. This often seems to not only make the outputs of the card its invisible to the windows kernel mixer but also change the sampling frequency of 44.1 sound card and stops work SpotOn.

  The second question I have is that the sampling frequency of default record when I record in edit mode is always 44.1 and if used it again change the map sound 44.1 and causes the same problems, I'd be very keen to know how to change this default to be 48 k if possible.

  Then.

  What I ultimatly looking is...

  1. a way to disable the ASIO drivers in hearing so that it is not only this option is available, and cannot use the Audition 3.0 Windows Sound Drivers.

  2. a way to make the sampling rate 48 k to stop people choosing 44.1 mistakenly when saving default record.

  Any help or advice that anyone can give would be much appreciated.

  Thanks in advance for your comments

  What other pilots ASIO sees your installation of 3 AA? If it's the Madiface one and you use only WDM drivers you can just uninstall the ASIO RME driver?

• Satellite M30, screen remains sometimes almost black after start up to several minutes

  Satellite M30, WinXP

  Sometimes, the screen remains dark, which means that you can see that a little bit news startup in the background and also the Office of windows, up to 2 minutes before suddenly become really turned on.
  Not even the latest version of the BIOS has a change. And what's weird, is it still does not appear. A non stop-and-shut-down-start sequence showed that it appears about 2 to 3 times.
  What could be the reason?

  Hi LeTom,

  did you use the os recovery toshiba or a clean install?

  It seems that the display shows a fault, then here are a few questions to play safely!

  There the failure also occure when the Red Toshiba logo appears shortly after starting start?

  If you are using a clean install of the OS, you need to do a complete new installation with the Toshiba Recovery CD (attention all data will be lost, make a backup).
  Then, the laptop is set back to the default settings of the software and if the failure occurred, you should check the material by a Toshiba service partner!

  I think that the M30 is still under warranty!

  Bossi Bye