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Thank you for such a wonderfull project
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Hi, please can you convert for c#?
Please, help me....
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Hi, and thank you for you perfect source code.
I want to make same program and I need some help of expert one.
my question is, why you shift 6 times the pSample[idx] in following code?
rm = abs(pSamples2[idx+1])>> 6
and when I modify this code for (channel == mono) or (bitpersample == 8),
the 6 should be modified with which number?
and how shoud I write my code to good working about all different values of
channel, samplepersec, and bitpersample parameters?
hello c++
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int nu = (int)(log(sampleCount)/log(2));
Error in this line above.
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Replace it by this line:
int nu = (int)log((double)sampleCount/log((double)2));
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Alternatively one might change the variable declaration of sampleCount to double from int and replace 2 with 2.000 for an implied type cast to double.
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hi, i got a program from codeproject which playes wave file with directsound. and i want to modify it and add spectrum support to it. but i got some questiones about it, can you help me resolve it?
question one:
the program open wave file and read wave file information to struct WAVEFORMATEX, the members of it like wFormatTag=1, nChannels=2, nSamplesPerSec=44100, nAvgBytesPerSec=176400, nBlockAlign=4, wBitsPerSample=16, cbSize=0. then program initial DirectSoundBuffer's size as 176400, we must set size to 176400, it can be set any number?
question two:
i add code to log DirectSoundBuffer's current play position and current write position to listbox control, the max value displayed in it which each of them may big than 176400, how can i get right start postion what i want to sample the buffer?
code like:
DWORD dwCurPlayPos, dwCurWritePos;
m_lpDSB->GetCurrentPosition(&dwCurPlayPos, &dwCurWritePos);
question three:
if i get right start postion of buffer, what right sample size may be suggest?
code modified by me can download here.
regards,
jacky_zz
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This is a usefull software
http://www.zeitnitz.de/Christian/Scope/Scope_en.html
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I'm an experienced and published telemarketing expert looking for a programmer as a partner. I'm interested in offering an anger detection service that would enable call centers to identify those calls involving angry customers.
Specifically, I need an application that can perform the following functions rapidly:
(1) automatically open one audio file after another
(2) analyze each file for mean pitch, maximum pitch variation, mean volume, maximum volume variation, and rate of speech
(3) Report the raw data in text format that can be downloaded into a Excel spreadsheet or other database
The programmer would receive 50% of gross revenue.
Are you interested in receiving a business proposal? Do you know someone else who might be?
Thanks you -
Stewart Rogers
stewart@100,000coldcalls.com
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I can do a swear detector?
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I have 400-500 recorded messages expressing anger. Can you use these to build an anlyzer that can identify the 10% of conversations most likely to express anger? Stewart
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Is it lawful to keep a database of customers calls? Does it not infringe on the Data Protection Act?
Also, have you got a psychologist or speech expert on-board to help you analyse and interpret data?
Without that, all you have is a database redundant data.
Good luck though.
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when i set sound player(Windows Media Player 9 up or Foobar2000) volumn from 100% to zero, the spectrum lower than what before i change system volumn, why??
modified on Tuesday, August 12, 2008 5:58 AM
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Hi,
This looks like a great project but I cannot find the source or even the application. Whats up?
Thanks,
Derek
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hi
i want to play two sound file same time and want to listen the sound of these file
on separtly on left and right speaker,means sound of one file should play on left
speaker and second file should play on right speaker
thank u in advance
malik
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Use waveOutWrite and put the data(average of left and right channel) of first file in left channel, put the data of second file in right channel. Refer to MSDN for details.
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Hi,
I'm getting this error: 'Error unable to set recording WAVEOUT device' when I click the Sample button. I've been able to follow what is happening but am unsure what it means.
In the SetInputDevice() function it has this if statement:
if(mixerGetLineControls(HMIXEROBJ)hMixer, &mxlc, MIXER_OBJECTF_HMIXER | MIXER_GETLINECONTROLSF_ONEBYTYPE) != MMSYSERR_NOERROR)
And that is where it fails returning a value of MIXERR_INVALCONTROL (1025) and I'm not sure what to do with that. My soundcard is a really basic setup. It is an ESS AUDIODRIVE.
Any help would be greatly appreciated.
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Hi,
The sound card sampling detection isn't perfect. Some sound cards may not have the WAVEOUT device, or have it named differently, or in your case do not have a mixer device.
This is the reason why it fails for you, short of getting some decient hardware, I can't offer you any software solution.
S.
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The mixer operation is not necessary, so just remove the line 349 "break;" in file "scope.cpp".
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23Dec2008
Greetings.
I have two questions.
1) I really have no idea how to recompile the source code
into an exe after editing out the "Break" line.
Might someone have a moment to tell me how to do that?
2) There is now a version 4 release of this audio scope.
(See the posting below about new version at SourceForge.)
Might someone know how to resolve this same error message
with the version 4 release? The v4 release seemss quite good.
I have three notebooks. One is 10 years old, and
I am trying to make it as multimedia capable as possible.
This 10 year old notebook gives me the
"Error unable to set recording WAVEOUT device" error message.
Any replies appreciated.
Thank you.
Regards,
AEN
Æ
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use many cpu percenge ,see winamp,cpu is 0
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Well its not hard to guess why, so I'll visit it for you:
1) Winamp does not sample from the audio card, the sample information is
store in the compressed audio stream, so there is no hardware API to the audio hardware.
2) They have probably used some optimised C/assembly routine to crank out the FFT for the audio data.
3) Its little video spectrum graphics are so tiny that video hardware can update this quickly.
Now looking at my example, its clear I've written it to show how to do it,
and how to do simply.
Most of the CPU/work is actually done on the graphic output, and again its done to keep things simple so you can understand it, which means using slow Windows GDI.
Look here for more better example of turning this simple example into something more demanding:
http://sourceforge.net/project/downloading.php?group_id=61001&use_mirror=optusnet&filename=audioscope_v4.zip&95969022[^]
And yes, it also does consume CPU. Again, mainly due to the graphics output and Windows GDI, of course your video hardware may vary.
Having said that the FFT and sampling routines are quite quick, even if they are not overly optimised.
Steve.
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SetInputDevice fails on the line
if (mixerGetLineInfo((HMIXEROBJ)hMixer, &mxl, MIXER_GETLINEINFOF_COMPONENTTYPE) != MMSYSERR_NOERROR)
under Windows Vista.
I'm not entirely sure what SetInputDevice is doing, but it looks like it's just making sure all the input settings are correct.
Luckily? the project seems to work fine with the calls in the IDC_RECORD case commented out...
Any comments or fixes?
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