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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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Hello from Portugal
I have modified the code of the application oscilloscope to make an application to generate blobs on a monitor for a dancing choreography with Aibos. I need to make a reference to your work because I used parts of it...What kind of reference can I make about you, have you ever published anything about the application oscilloscope, a scientific article for example, can you give me more information’s than the ones in the site??Pease send me an answer as fast as you can, because a need to publish my article…sorry for any problem…and thank you for the help that you gave me, with the code of you application.
bite@portugalmail.pt
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Hi guys.
Sorry for the barrage of emails lately but I really do need help getting this done so that I can move on to other parts of my project. If you could spare a minute, I would really appreciate it, thank you.
For those of you who don't already know my problem- I am trying to incorporate this (mainly the input magnitude and FFT spectrum results) into an application I have already implemented (in a VC++ Express Edition Windows Form). I have split the Steven's code into three main sub-routines/functions- StartRec, StopRec and the other is a callback function WaveInProc() (associated with WaveInOpen).
I don't know if this approach is "correct" or if it will work but I am now stuck at the beginning- the StartRec function! It seems that I have an <undefined value=""> for hMixer in the SetInDevice subroutine.
HMIXER hMixer = NULL;
int inDevIdx = -1;
if (mixerOpen(&hMixer, 0, 0, NULL, MIXER_OBJECTF_MIXER) != MMSYSERR_NOERROR)
{
return FALSE;
}
However, when I debug Steven's original code, hMixer first had a value of 0xcccccccc then "0x00000000 {unused = ???}" (whilst I got a <undefined value="">). I really don't know what is happening here; the exact same code is producing these two different results. The answer might be really simple but I really don;t know it and could find much on the internet that explain this difference.
Please help if you have the time. Thank you very very much.
Regards,
Jeff
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this call :
HMIXER hMixer = NULL;
int inDevIdx = -1;
if (mixerOpen(&hMixer, 0, 0, NULL, MIXER_OBJECTF_MIXER) != MMSYSERR_NOERROR)
{
return FALSE;
}
this is correct, it opens a mixer device, if it fails then you need to check the return value:
Return Values
Returns MMSYSERR_NOERROR if successful or an error otherwise. Possible error values include the following:
Value Description
MMSYSERR_ALLOCATED The specified resource is already allocated by the maximum number of clients possible.
MMSYSERR_BADDEVICEID The uMxId parameter specifies an invalid device identifier.
MMSYSERR_INVALFLAG One or more flags are invalid.
MMSYSERR_INVALHANDLE The uMxId parameter specifies an invalid handle.
MMSYSERR_INVALPARAM One or more parameters are invalid.
MMSYSERR_NODRIVER No mixer device is available for the object specified by uMxId. Note that the location referenced by uMxId will also contain the value – 1.
MMSYSERR_NOMEM Unable to allocate resources.
the hMixer having a value of 0xcccccccc initially is the debugger setting this to track possible memory leaks. After passing of the
handle/variable by address, it looks like mixerOpen() is setting the hMixer variable to NULL due to an error.
Review the mixerOpen() API function on www.msdn.com or
http://msdn2.microsoft.com/en-us/library/ms712134.aspx
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Hi Steven.
Thank you very much for your quick reply, really appreciate it. I am also really sorry about my late response, been swamped with other work lately.
I think I just know too little on this. I did review msdn and various sources in order to study your code and I think I understand it. But, maybe because it is based on a dialogbox, I just can't seem to incorporate it into what I already have.
"All" I need are the VALUES of the microphone level (MaxLeft and MaxRight in your code) and the spectrum (pMags[idx]). I have been tweaking and studying your code (and the relevant API calls) for weeks and I still can't get past the HMIXER bit, not to mention obtaining the values. I really don't to give this up but I really don't have that much time to spare for just this component. If it is not too much too ask, could you give me some pointers in how I can approach this?
Thanks again.
Regards,
Jeff
-- modified at 6:42 Friday 15th June, 2007
These are the parts of Steven's code I'm using:
// Audio Scope Written By Steven De Toni
#include "stdafx.h"
#include "aud.h"
#include <stdio.h>
#include <stdarg.h>
#include <windows.h>
#include <commctrl.h>
#include <mmsystem.h>
#include "FFTransform.h"
//#include "resource.h"
// --- Global Variables ---
HINSTANCE HInst = NULL;
void CALLBACK waveInProc(HWAVEIN hWaveIn, UINT uMsg, DWORD dwInstance,
DWORD dwParam1, DWORD dwParam2);
void OutputDebugMsg (const char* pszFormat, ...)
{
char buf[1024];
va_list arglist;
va_start(arglist, pszFormat);
vsprintf(buf, pszFormat, arglist);
va_end(arglist);
strcat(buf, "\n");
OutputDebugString(buf);
}
void OutputMsgBox (const char* pszFormat, ...)
{
char buf[1024];
va_list arglist;
va_start(arglist, pszFormat);
vsprintf(buf, pszFormat, arglist);
va_end(arglist);
strcat(buf, "\n");
MessageBox (NULL, buf, "", MB_ICONSTOP);
}
/*********************** PrintWaveErrorMsg() **************************
* Retrieves and displays an error message for the passed Wave In error
* number. It does this using mciGetErrorString().
*************************************************************************/
void PrintWaveErrorMsg(DWORD err, TCHAR * str)
{
char buffer[128];
OutputMsgBox ("ERROR 0x%08X: %s\r\n", err, str);
if (mciGetErrorString(err, &buffer[0], sizeof(buffer)))
{
OutputMsgBox ("%s\r\n", &buffer[0]);
}
else
{
OutputMsgBox ("0x%08X returned!\r\n", err);
}
}
/*
What a long winded way to select recording input!
MIXERLINE_COMPONENTTYPE_SRC_ANALOG
MIXERLINE_COMPONENTTYPE_SRC_AUXILIARY
MIXERLINE_COMPONENTTYPE_SRC_COMPACTDISC
MIXERLINE_COMPONENTTYPE_SRC_DIGITAL
MIXERLINE_COMPONENTTYPE_SRC_LINE
MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE
MIXERLINE_COMPONENTTYPE_SRC_PCSPEAKER
MIXERLINE_COMPONENTTYPE_SRC_SYNTHESIZER
MIXERLINE_COMPONENTTYPE_SRC_TELEPHONE
MIXERLINE_COMPONENTTYPE_SRC_UNDEFINED
MIXERLINE_COMPONENTTYPE_SRC_WAVEOUT
*/
BOOL SetInputDevice (unsigned int inDev)
{
HMIXER hMixer = NULL;
int inDevIdx = -1;
if ((mixerOpen(&hMixer, 0, 0, NULL, MIXER_OBJECTF_MIXER)) != MMSYSERR_NOERROR)
{
return FALSE;
}
// get dwLineID
MIXERLINE mxl;
mxl.cbStruct = sizeof(MIXERLINE);
mxl.dwComponentType = MIXERLINE_COMPONENTTYPE_DST_WAVEIN;
if (mixerGetLineInfo((HMIXEROBJ)hMixer, &mxl, MIXER_OBJECTF_HMIXER | MIXER_GETLINEINFOF_COMPONENTTYPE) != MMSYSERR_NOERROR)
{
mixerClose (hMixer);
return FALSE;
}
// get dwControlID
MIXERCONTROL mxc;
MIXERLINECONTROLS mxlc;
DWORD dwControlType = MIXERCONTROL_CONTROLTYPE_MIXER;
mxlc.cbStruct = sizeof(MIXERLINECONTROLS);
mxlc.dwLineID = mxl.dwLineID;
mxlc.dwControlType = dwControlType;
mxlc.cControls = 0;
mxlc.cbmxctrl = sizeof(MIXERCONTROL);
mxlc.pamxctrl = &mxc;
if (mixerGetLineControls((HMIXEROBJ)hMixer, &mxlc, MIXER_OBJECTF_HMIXER | MIXER_GETLINECONTROLSF_ONEBYTYPE) != MMSYSERR_NOERROR)
{
// no mixer, try MUX
dwControlType = MIXERCONTROL_CONTROLTYPE_MUX;
mxlc.cbStruct = sizeof(MIXERLINECONTROLS);
mxlc.dwLineID = mxl.dwLineID;
mxlc.dwControlType = dwControlType;
mxlc.cControls = 0;
mxlc.cbmxctrl = sizeof(MIXERCONTROL);
mxlc.pamxctrl = &mxc;
if (mixerGetLineControls((HMIXEROBJ)hMixer, &mxlc, MIXER_OBJECTF_HMIXER | MIXER_GETLINECONTROLSF_ONEBYTYPE) != MMSYSERR_NOERROR)
{
mixerClose (hMixer);
return FALSE;
}
}
if (mxc.cMultipleItems <= 0)
{
mixerClose (hMixer);
return FALSE;
}
// get the index of the inDevice from available controls
MIXERCONTROLDETAILS_LISTTEXT* pmxcdSelectText = new MIXERCONTROLDETAILS_LISTTEXT[mxc.cMultipleItems];
if (pmxcdSelectText != NULL)
{
MIXERCONTROLDETAILS mxcd;
mxcd.cbStruct = sizeof(MIXERCONTROLDETAILS);
mxcd.dwControlID = mxc.dwControlID;
mxcd.cChannels = 1;
mxcd.cMultipleItems = mxc.cMultipleItems;
mxcd.cbDetails = sizeof(MIXERCONTROLDETAILS_LISTTEXT);
mxcd.paDetails = pmxcdSelectText;
if (mixerGetControlDetails ((HMIXEROBJ)hMixer, &mxcd, MIXER_OBJECTF_HMIXER | MIXER_GETCONTROLDETAILSF_LISTTEXT) == MMSYSERR_NOERROR)
{
// determine which controls the inputDevice source line
DWORD dwi;
for (dwi = 0; dwi < mxc.cMultipleItems; dwi++)
{
// get the line information
MIXERLINE mxl;
mxl.cbStruct = sizeof(MIXERLINE);
mxl.dwLineID = pmxcdSelectText[dwi].dwParam1;
if (mixerGetLineInfo ((HMIXEROBJ)hMixer, &mxl, MIXER_OBJECTF_HMIXER | MIXER_GETLINEINFOF_LINEID) == MMSYSERR_NOERROR && mxl.dwComponentType == inDev)
{
// found, dwi is the index.
inDevIdx = dwi;
// break;
}
}
}
delete []pmxcdSelectText;
}
if (inDevIdx < 0)
{
mixerClose (hMixer);
return FALSE;
}
// get all the values first
MIXERCONTROLDETAILS_BOOLEAN* pmxcdSelectValue = new MIXERCONTROLDETAILS_BOOLEAN[mxc.cMultipleItems];
if (pmxcdSelectValue != NULL)
{
MIXERCONTROLDETAILS mxcd;
mxcd.cbStruct = sizeof(MIXERCONTROLDETAILS);
mxcd.dwControlID = mxc.dwControlID;
mxcd.cChannels = 1;
mxcd.cMultipleItems = mxc.cMultipleItems;
mxcd.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
mxcd.paDetails = pmxcdSelectValue;
if (mixerGetControlDetails((HMIXEROBJ)hMixer, &mxcd, MIXER_OBJECTF_HMIXER | MIXER_GETCONTROLDETAILSF_VALUE) == MMSYSERR_NOERROR)
{
// ASSERT(m_dwControlType == MIXERCONTROL_CONTROLTYPE_MIXER || m_dwControlType == MIXERCONTROL_CONTROLTYPE_MUX);
// MUX restricts the line selection to one source line at a time.
if (dwControlType == MIXERCONTROL_CONTROLTYPE_MUX)
{
ZeroMemory(pmxcdSelectValue, mxc.cMultipleItems * sizeof(MIXERCONTROLDETAILS_BOOLEAN));
}
// Turn on this input device
pmxcdSelectValue[inDevIdx].fValue = 0x1;
mxcd.cbStruct = sizeof(MIXERCONTROLDETAILS);
mxcd.dwControlID = mxc.dwControlID;
mxcd.cChannels = 1;
mxcd.cMultipleItems = mxc.cMultipleItems;
mxcd.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
mxcd.paDetails = pmxcdSelectValue;
if (mixerSetControlDetails ((HMIXEROBJ)hMixer, &mxcd, MIXER_OBJECTF_HMIXER | MIXER_SETCONTROLDETAILSF_VALUE) != MMSYSERR_NOERROR)
{
delete []pmxcdSelectValue;
mixerClose (hMixer);
return FALSE;
}
}
delete []pmxcdSelectValue;
}
mixerClose (hMixer);
return TRUE;
}
// ***********************************************************************************
#define SPECSCOPEWIDTH 10
static BOOL inRecord = FALSE;
static HWAVEIN waveInHandle = NULL;
static WAVEHDR waveHeader[2];
static WAVEFORMATEX waveFormat;
static FFTransform* pFFTrans = NULL;
static FFTransform* pFFTransStereo = NULL;
static SampleIter* pSampleIter = NULL;
static RECT drawArea;
static HDC HdblDC = NULL;
static HBITMAP HdblOldBitmap = NULL;
static int rps = 1;
///////////////////////////////////////////////////////////////////////////////
Caud::Caud(){}
void Caud::StopRec(){
// Stop recording and tell the driver to unqueue/return all of our WAVEHDRs.
// The driver will return any partially filled buffer that was currently
// recording. Because we use waveInReset() instead of waveInStop(),
// all of the other WAVEHDRs will also be returned via MM_WIM_DONE too
waveInReset(waveInHandle);
waveInUnprepareHeader (waveInHandle, &waveHeader[0], sizeof(WAVEHDR));
waveInUnprepareHeader (waveInHandle, &waveHeader[1], sizeof(WAVEHDR));
waveInClose(waveInHandle);
VirtualFree(waveHeader[0].lpData, (waveHeader[0].dwBufferLength << 1), MEM_RELEASE);
inRecord = FALSE;
SetInputDevice (MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE);
if (pFFTrans != NULL)
{
delete pFFTrans;
pFFTrans = NULL;
}
if (pFFTransStereo != NULL)
{
delete pFFTransStereo;
pFFTransStereo = NULL;
}
if (pSampleIter != NULL)
{
delete pSampleIter;
pSampleIter = NULL;
}
}
////////////////////////////////////////////////////////////////////////////////////////////////////
void Caud::StartRec() {
//InitCommonControls ();
while (1) {
// Set the recording input ...
//if (SetInputDevice (MIXERLINE_COMPONENTTYPE_SRC_WAVEOUT) == FALSE)
//{
if (SetInputDevice (MIXERLINE_COMPONENTTYPE_SRC_ANALOG) == FALSE)
{
if (SetInputDevice (MIXERLINE_COMPONENTTYPE_SRC_LAST) == FALSE)
{
OutputMsgBox ("Error unable to set recording WAVEOUT device");
break;
}
}
//}
MMRESULT err;
// Clear out both of our WAVEHDRs. At the very least, waveInPrepareHeader() expects the dwFlags field to be cleared
ZeroMemory(&waveHeader[0], sizeof(WAVEHDR) * 2);
// Initialize the WAVEFORMATEX for 16-bit, 44KHz, stereo. That's what I want to record
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nChannels = 2;
waveFormat.nSamplesPerSec = 44100;
waveFormat.wBitsPerSample = 16;
waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;
// Open the default WAVE In Device, specifying my callback. Note that if this device doesn't
// support 16-bit, 44KHz, stereo recording, then Windows will attempt to open another device
// that does. So don't make any assumptions about the name of the device that opens. After
// waveInOpen returns the handle, use waveInGetID to fetch its ID, and then waveInGetDevCaps
// to retrieve the actual name
if ((err = waveInOpen(&waveInHandle, WAVE_MAPPER, &waveFormat, (DWORD)waveInProc, 0, CALLBACK_FUNCTION )) != 0)
{
PrintWaveErrorMsg (err, "Can't open WAVE In Device!");
break;
}
// Allocate, prepare, and queue two buffers that the driver can use to record blocks of audio data.
// (ie, We're using double-buffering. You can use more buffers if you'd like, and you should do that
// if you suspect that you may lag the driver when you're writing a buffer to disk and are too slow
// to requeue it with waveInAddBuffer. With more buffers, you can take your time requeueing
// each).
//
// I'll allocate 2 buffers large enough to hold 2 seconds worth of waveform data at 44Khz. NOTE:
// Just to make it easy, I'll use 1 call to VirtualAlloc to allocate both buffers, but you can
// use 2 separate calls since the buffers do NOT need to be contiguous. You should do the latter if
// using many, large buffers
waveHeader[1].dwBufferLength = waveHeader[0].dwBufferLength = 512;
if (!(waveHeader[0].lpData = (char*)VirtualAlloc(0, (waveHeader[0].dwBufferLength << 1), MEM_COMMIT, PAGE_READWRITE)))
{
OutputMsgBox ("ERROR: Can't allocate memory for WAVE buffer!\n");
waveInClose(waveInHandle);
break;
}
// Fill in WAVEHDR fields for buffer starting address. We've already filled in the size fields above */
waveHeader[1].lpData = waveHeader[0].lpData + waveHeader[0].dwBufferLength;
// Leave other WAVEHDR fields at 0
// Prepare the 2 WAVEHDR's
if ((err = waveInPrepareHeader(waveInHandle, &waveHeader[0], sizeof(WAVEHDR))))
{
waveInClose(waveInHandle);
OutputMsgBox ("Error preparing WAVEHDR 1! -- %08X\n", err);
VirtualFree (waveHeader[0].lpData, (waveHeader[0].dwBufferLength << 1), MEM_RELEASE);
break;
}
if ((err = waveInPrepareHeader(waveInHandle, &waveHeader[1], sizeof(WAVEHDR))))
{
waveInClose(waveInHandle);
OutputMsgBox ("Error preparing WAVEHDR 2! -- %08X\n", err);
VirtualFree (waveHeader[0].lpData, (waveHeader[0].dwBufferLength << 1), MEM_RELEASE);
break;
}
// Queue first WAVEHDR (recording hasn't started yet)
if ((err = waveInAddBuffer(waveInHandle, &waveHeader[0], sizeof(WAVEHDR))))
{
waveInClose(waveInHandle);
OutputMsgBox ("Error queueing WAVEHDR 1! -- %08X\n", err);
VirtualFree (waveHeader[0].lpData, (waveHeader[0].dwBufferLength << 1), MEM_RELEASE);
break;
}
// Queue second WAVEHDR
if ((err = waveInAddBuffer(waveInHandle, &waveHeader[1], sizeof(WAVEHDR))))
{
waveInClose(waveInHandle);
OutputMsgBox ("Error queueing WAVEHDR 2! -- %08X\n", err);
VirtualFree (waveHeader[0].lpData, (waveHeader[0].dwBufferLength << 1), MEM_RELEASE);
break;
}
// Start recording
if ((err = waveInStart(waveInHandle)))
{
OutputMsgBox ("Error starting record! -- %08X\n", err);
waveInClose(waveInHandle);
VirtualFree (waveHeader[0].lpData, (waveHeader[0].dwBufferLength << 1), MEM_RELEASE);
break;
}
// prepare the DSP processing objects
pFFTrans = new FFTransform (waveFormat.nSamplesPerSec, waveHeader[0].dwBufferLength/(waveFormat.nChannels * (waveFormat.wBitsPerSample/8)));
pFFTransStereo = new FFTransform (waveFormat.nSamplesPerSec, waveHeader[0].dwBufferLength/(waveFormat.wBitsPerSample/8));
pSampleIter = new SampleIter();
inRecord = TRUE;
break;
}
}
/////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
void CALLBACK waveInProc(HWAVEIN hWaveIn, UINT uMsg, DWORD dwInstance,
DWORD dwParam1, DWORD dwParam2) {
if (uMsg == WIM_DATA){
WAVEHDR* pHeader = (WAVEHDR*) dwParam1;
static int maxLeftSave = 0;
static int maxRightSave = 0;
static int refresh = 0;
refresh++;
if (pHeader->dwBytesRecorded > 0)
{
// OutputDebugMsg ("Recorded %d bytes\n", pHeader->dwBytesRecorded);
int idx = 0;
short int* pSamples = (short int*)pHeader->lpData;
int sampleCnt = pHeader->dwBytesRecorded/(sizeof(short int));
// ------------------------------------------------------------------
// ******* Calculate Power Meter *******
int lm = 0, rm = 0, maxLeft = 0, maxRight = 0;
for (idx = 0; idx < sampleCnt; idx += 2)
{
if ((lm = abs(pSamples[idx]) >> 6) > maxLeft)
maxLeft = lm;
if ((rm = abs(pSamples[idx+1]) >> 6) > maxRight)
maxRight = rm;
}
if (maxLeft < maxLeftSave)
{
if ((maxLeft = maxLeftSave - 4) < 0)
maxLeft = 0;
}
if (maxRight < maxRightSave)
{
if ((maxRight = maxRightSave - 4) < 0)
maxRight = 0;
}
maxLeftSave = maxLeft;
maxRightSave = maxRight;
}
if (inRecord) // Are we still recording?
{
// Yes. Now we need to requeue this buffer so the driver can use it for another block of audio
// data. NOTE: We shouldn't need to waveInPrepareHeader() a WAVEHDR that has already been prepared once
waveInAddBuffer(waveInHandle, pHeader, sizeof(WAVEHDR));
}
}
}
////////////////////////////////////////////////////////////////
I am only aiming to get the input value (MaxLeft and MaxRight) now. First I call StartRec() at the beginning and StopRec() at the end of my own program. I am probably something stupidly wrong here. Anyway, any help would be MUCH MUCH appreciated. Thank you lots.
Jeff
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Hi Steven!
Why can not download the source file form http://www.codeproject.com/audio/oscilloscope/oscilloscope_src.zip ?
Could you send me to the e-mail "david_hsieh_173@hotmail.com"
Thanks!
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Hi Steven!
Cool project! And very useful as well. I have looked at your source code and am trying to utilize it for my own project (hope you don't mind) but I am having trouble reorganizing it. I was wondering if you or indeed any expert who knows this code and C++ well could help me?
What I am trying to do is basically to incorporate the FFT and oscilloscope (i.e. the amplitude) results into a C++ project (with a Windows Form) I have already implemented. Unfortunately, I am pretty incompetent and unable to transfer the code. I am a bit stuck on the all-important BOOL CALLBACK Scope call. From what I understand, this deals with the messages sent by the dialog box and then proceeds to perform a specific function (e.g. start recording, stop recording, process audio messages). As I am not using a dialog box for my current project, I am going with the approach of call these functions myself within my main loop.
Where I am really lost is the part where it processes the audio messages ("Process Audio Messages"). I haven't got a clue on what to substitute the "WAVEHDR* pHeader = (WAVEHDR*) lParam" for. From my limited knowledge (and some research), I assume that the lParam is a part of the data that is sent by the dialog box but since I am not using a dialog box and want to continually process the audio messages, I am not sure how to change pHearder and its attributes, such as dwBytesRecorded.
I would really appreciate any help or advice. Thank you very very much.
Regards,
Jeff
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jeffhcliu wrote: Where I am really lost is the part where it processes the audio messages ("Process Audio Messages"). I haven't got a clue on what to substitute the "WAVEHDR* pHeader = (WAVEHDR*) lParam" for. From my limited knowledge (and some research), I assume that the lParam is a part of the data that is sent by the dialog box but since I am not using a dialog box and want to continually process the audio messages, I am not sure how to change pHearder and its attributes, such as dwBytesRecorded.
I think you need to study the code more closely or in conjunction with the WIN32 documentation from the microsoft developer web site. You cannot sample the audio stream constantly as you do not have access to the sounnd hardware, only the sound drivers/API. So to process sound information, we define the sample information we need, 16bit, 22khz, 1k buffer etc. The sound driver will then DELIVER that information back to the program via what is known as a callback. This example program uses the Windows message system as a notification to process the sound buffer. You can also implement a function callback if your program is not associated within Windows messages.
Look at the code here:
http://sourceforge.net/project/showfiles.php?group_id=61001&package_id=226483
for a more complete example. How to implement thread processing, and function callback.
The lParam is really just a 32bit variable we use to pass information around within Win32 API calls. It could be any 32bit value in relation to a program, its commonly used to store a pointer/address value the points to additional program data. In your case the pointer to a WAVEHDR structure, the lParam value is type case from a LPARAM type (32bit long) to a WAVEHDR pointer so the program understands data/structure at that memory location.
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Hi Steven.
Thank you so much for your quick reply. Much appreciated. I guess I do superficially understand what the code does and have studied it several times but the problem (I think) I am having is that I can't see a way of utilizing it in my program (due to my lack of knowledge on programming).
I have my own program which does a bunch of things within an infinite loop. I would like to add the capabilities of your program to mine, i.e. being able to sample from a microphone and doing FFT analysis on it. In other words, as soon as I clicked on a button, I would want start the sound sampling (along with all the other stuff within that infinite loop) and perform FFT, etc. Then when I click on another button, I will break that loop as well as stopping the sampling.
Now, as your program creates a dialog box and is based on Windows Messages, I am not really sure how to translate that with what I know. In the high likelihood of sounding extremely stupid, I have changed the WM_COMMAND case into StartRecord and StopRecord "void" subroutines which will be called when certain buttons (on a Windows Form) are clicked. Now, I am searching for a way of translating the MM_WIM_DATA part into a subroutine (ProcessAudio), or a function callback as you have kindly suggested.
But I really don't know how to do this. I think I am facing two main problems at this point (based on my approach). If I don't use Windows Messages, what am I going to pass as the lParam value to ProcessAudio (as the MM_WIM_DATA case uses it heavily)? The other problem is that I do not want ProcessAudio to be processing all the time (i.e. "in an infinite loop of its own", in my words) as there will be other stuff that will need to be processed within my own main infinite loop, of which ProcessAudio will be only a part of. I have read up on multi-threading as well as callbacks but I still don't know how to proceed.
I hope you understand what I have just said and what I am trying to do and please forgive me if I am doing something very stupid as I really don't have that much programming experience.
A million thanks.
Regards,
Jeff
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hi
do you have any idea how to change the program so that it works for a sound card (like realtek HD) which has different mixers for input and output?
(actually it has 3:
0: Realtek HD Audio Output
1: Realtek HD Digital Input
2: Realtek HD Audio Input
)
so the mixerOpen with the device id "0" doesn't work anymore
i am getting the default wavein handle, then open the mixer using the waveinhandle as a parameter
with the mixerGetLineInfo i am getting 4 sources with the wavein destination
( 0: CD Volume;
1: Mic Volume
2: Line Volume
3: Stereo Mix
after that i get stuck
the next calls to mixerGetLineControls return error code 1025 which is MIXERR_INVALCONTROL
does it mean that i don't have any mix/mux controls. why do i need those?
i am sure i didn't understand well how the mixer api works.
if you have any sugestions it would be very nice
also another thing i noticed on a sound card with a singe mixer is that the spectrum analizer depends on the sound volume. if i change the music volume the spectrum analizer changes with it.
i was wondering what winamp and other applications use, because they seem to not care about the sound volume. the spectrum analizer remains always the same even if the sound is muted.
thanks in advance
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hi Steven,
Really good project. Is there anyway you could send me flowchart/psuedo code for this project. Thanks in advance.
With Regards,
G.Vidyadhar
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Thank for your sharing, here i would like to ask a question, how to get the amplitude and frequecy from microphone?
i just want to differentiate the noisy level of input data only,and get the digital value.
thank
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Hello Steve,
with your code I learned a lot about mixer, sound and so on. Thanks too much.
1) What I learned is then used in my application which has to "monitor" a microphone. Unfortunatelly I do not understand why my application stops to receive MM_WIM_DATA message, right after a mouse click has been done on a CListCtrl control or a CButton. This doesn't happens even if I do any other operation inside my application: for example start recording. I do not receive any error messages (each function has the result controlled) and I do not receive any other MM_WIM_XXX like MM_WIM_CLOSE.
When this happens, I need to call again the "waveInAddBuffer" and the "waveInStart": then every thing starts to work again (until a next mouse click).
Do you have any suggestion?
2) I need to put a "parametric equalizer" inside my application. Info collected until now, tell me that I need FFT transformation to have that equalizer? Any help or any suggestion.
Thanks again for your work and in advance for your help.
GianniGP
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Hello Steve, too busy?
Well I found that there is no solution on the MM_WIM_DATA problem. The only thing that works, is to open the waveInOpen() using a CALLBACK_FUNCTION and the relative waveInProc(). In this way never happens that the buffer stops to get back to the application.
Still I'm asking you the second question of my message: the parametri equalizer.
Thanks and regards.
GianniGP
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Hi Gianni.
Could you please help me in converting this code for use in a Windows Form already implemented? Thank you.
As explained in my earlier post, I have already implemented a Visual C++ Form with an infinite loop that does a bunch of things. Now I would like to retrieve microphone input amplitude as well as FFT data utilizing Steven's code.
Do you have any idea on how to do this? I am just too stupid to change this dialog box and message scheme into the functions I want. I have already done quite a bit of research and reasonably understand how Steven's program works, using dialogbox and various messages, but still can't get my head around converting them. How did you convert them into callback FUNCTIONS and what should I use for parameters such as (WAVEHDR*) lParam?
One thing to note is that I do not want ONLY this program/features to be run as there will also be other things that have to be processed as well in the infinite loop.
Any help or direction will be much appreciated. Thank you very much.
Regards,
Jeff
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Hi,
I want to convert your code in C#. But i have not worked with C++ so can you please guide me, what to do and where to start using your code?
Help me out please,
Thanks,
Mahesh Devjibhai Dhola
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Sorry, I have no idea, have not touch C#...
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you will definitely get some good converter.. freeware. over the internet
Regards,
Tauhid Shaikh
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i can help if you are still interested - i see its pretty dated so maybe already found solution?
-O
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Hello,
thanks for this article/source code.
I noticed the following:
When you calculate the spectrum values using FFT, you calculate for left, right and both. There you have a call
float* pMags = pFFTrans->fftGetMagArray();
where in fact it should be
float* pMags = pFFT->fftGetMagArray(); to ensure that you use pFFTTransStereo for the third case.
I hope my assumption is correct.
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