[Stk] RtApiDummy: This class provides no functionality.

Wed Sep 10 13:55:54 PDT 2008

Dear Gary and Steven;

 Thank you very much for the advice.

I compiled RtAudio-4.0.4 and STK-4.3.1 using ./configure --with-core
and I used the preprocessor as atthched in the program:

/******************************************/
/*
 audioprobe.cpp
 by Gary P. Scavone, 2001

 Probe audio system and prints device info.
 */
/******************************************/
#ifndef __MACOSX_CORE__
#define __MACOSX_CORE__
#endif

#include "RtAudio.h"
#include <iostream>
#include <map>

#include <CoreAudio/AudioHardware.h>

/*
 typedef char  MY_TYPE;
 #define FORMAT RTAUDIO_SINT8

 typedef signed short  MY_TYPE;
 #define FORMAT RTAUDIO_SINT16

 typedef signed long  MY_TYPE;
 #define FORMAT RTAUDIO_SINT24

 typedef signed long  MY_TYPE;
 #define FORMAT RTAUDIO_SINT32
 */

typedef float  MY_TYPE;
#define FORMAT RTAUDIO_FLOAT32

/*
 typedef double  MY_TYPE;
 #define FORMAT RTAUDIO_FLOAT64
 */

// Platform-dependent sleep routines.
#if defined( __WINDOWS_ASIO__ ) || defined( __WINDOWS_DS__ )
#include <windows.h>
#define SLEEP( milliseconds ) Sleep( (DWORD) milliseconds )
#else // Unix variants
#include <unistd.h>
#define SLEEP( milliseconds ) usleep( (unsigned long) (milliseconds * 1000.0) )
#endif

struct InputData {
	MY_TYPE* buffer;
	unsigned long bufferBytes;
	unsigned long totalFrames;
	unsigned long frameCounter;
	unsigned int channels;
};

// Interleaved buffers
int input( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
		  double streamTime, RtAudioStreamStatus status, void *data )
{
	InputData *iData = (InputData *) data;

	// Simply copy the data to our allocated buffer.
	unsigned int frames = nBufferFrames;
	if ( iData->frameCounter + nBufferFrames > iData->totalFrames ) {
		frames = iData->totalFrames - iData->frameCounter;
		iData->bufferBytes = frames * iData->channels * sizeof( MY_TYPE );
	}

	unsigned long offset = iData->frameCounter * iData->channels;
	memcpy( iData->buffer+offset, inputBuffer, iData->bufferBytes );
	iData->frameCounter += frames;

	if ( iData->frameCounter >= iData->totalFrames ) return 2;
	return 0;
}

int main( int argc, char *argv[] )
{
	unsigned int channels = 1, fs = 44100, bufferFrames, device = 0, offset = 0;
	double time = 2.0;
	FILE *fd;

	RtAudio adc;
	if ( adc.getDeviceCount() < 1 ) {
		std::cout << "\nNo audio devices found!\n";
		exit( 1 );
	}

	//
   time =  3.0;

	// Let RtAudio print messages to stderr.
	adc.showWarnings( true );

	// Set our stream parameters for input only.
	bufferFrames = 512;
	RtAudio::StreamParameters iParams;
	iParams.deviceId = device;
	iParams.nChannels = channels;
	iParams.firstChannel = offset;

	InputData data;
	data.buffer = 0;
	try {
		adc.openStream( NULL, &iParams, FORMAT, fs, &bufferFrames, &input,
(void *)&data );
	}
	catch ( RtError& e ) {
		std::cout << '\n' << e.getMessage() << '\n' << std::endl;
		goto cleanup;
	}

	data.bufferBytes = bufferFrames * channels * sizeof( MY_TYPE );
	data.totalFrames = (unsigned long) (fs * time);
	data.frameCounter = 0;
	data.channels = channels;
	unsigned long totalBytes;
	totalBytes = data.totalFrames * channels * sizeof( MY_TYPE );

	// Allocate the entire data buffer before starting stream.
	data.buffer = (MY_TYPE*) malloc( totalBytes );
	if ( data.buffer == 0 ) {
		std::cout << "Memory allocation error ... quitting!\n";
		goto cleanup;
	}

	try {
		adc.startStream();
	}
	catch ( RtError& e ) {
		std::cout << '\n' << e.getMessage() << '\n' << std::endl;
		goto cleanup;
	}

	std::cout << "\nRecording for " << time << " seconds ... writing file
'record.raw' (buffer frames = " << bufferFrames << ")." << std::endl;
	while ( 1 ) {
		SLEEP( 100 ); // wake every 100 ms to check if we're done
		if ( adc.isStreamRunning() == false ) break;
	}

	// Now write the entire data to the file.
	fd = fopen( "record.wav", "wb" );
	fwrite( data.buffer, sizeof( MY_TYPE ), data.totalFrames * channels, fd );
	fclose( fd );

cleanup:
	if ( adc.isStreamOpen() ) adc.closeStream();
	if ( data.buffer ) free( data.buffer );

	return 0;
}

I tried audioprobe.cpp, some examples in the STK tutorial, also in my
task which I need to rewrite for spoken command interface for speech
feature extraction in our project. The preprocessor I add is as
follows:

But I have no success yet while trying this but the output is

RtApiDummy: This class provides no functionality.

RtApi::openStream: output device parameter value is invalid.

// __MACOSX_CORE__: OS X specific preprocessor
#ifndef __MACOSX_CORE__
#define __MACOSX_CORE__
#endif

#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) ||
defined(__WINDOWS_MM__)
#define __OS_WINDOWS__
#define __STK_REALTIME__
#elif defined(__LINUX_OSS__) || defined(__LINUX_ALSA__) ||
defined(__LINUX_JACK__)
#define __OS_LINUX__
#define __STK_REALTIME__
#elif defined(__IRIX_AL__)
#define __OS_IRIX__
#define __STK_REALTIME__
#elif defined(__MACOSX_CORE__)
#define __OS_MACOSX__
#define __STK_REALTIME__
#endif

#include <CoreAudio/AudioHardware.h>

_rec_status(false),
_adc(new RtAudio(RtAudio::MACOSX_CORE)),//_ASIO)),
_sampleRate(44100),//Hz
_bufferFrames(1024),//number of samples per cycle
_DCcoeffs ( *(new wxClientDC (panel_FT) ) ),
_scale_x (512), _scale_y(512), // dimension of panel_FT
_x ( new Signal), // needed for callbackfunction - RtAudio
_calls (0),//callback-counter
_cost (Entropy),
_costpar (0.1),
_fixed_analyze_size (8192),
_fb_counter (0),
_fixed_buffer( new real [_fixed_analyze_size] ),

_F (new FEATEXTR (*this, _fixed_analyze_size, 6U, 64U) )

{

	assert(_adc);
	assert(_x);
	assert(_fixed_buffer);
	_x->RtAudioInterface(_fixed_buffer, _fixed_analyze_size);

	_COUT.open("/Users/Desktop/tuklwx/recordtestdata");
	std::cout << _x << std::endl;

	_adc-> getCurrentApi();
	if ( _adc -> getDeviceCount() < 1 ) {
		wxMessageBox ( _("No audio devices found!"));

		button_rec -> Enable(false);

unsigned deviceId = 1;
	_deviceinfo=_adc->getDeviceInfo(deviceId);// should be US-1641 for an ASIO api

	_parameters.deviceId = deviceId;
	_parameters.nChannels = 1;//_deviceinfo.inputChannels;//should be == 16
	_parameters.firstChannel = 0;

	//statusBar->SetStatusText(_deviceinfo.name, 1);

	_adc-> getCurrentApi();

Unfortunately, I could not proceed further to start the next task
involved here while using AudioInterface.  I do not how to solve this
problem. I will appreciate any recommendations that help me to
proceed.

Thank you and look forward to hearing from you,
Jui

On Wed, Sep 10, 2008 at 1:18 AM, alpana jui <lpnjui at googlemail.com> wrote:
> Dear Everybody,
>
>
>  I tried to set  MACOSX_CORE but I have the same output  as attached
> in all the RtAudio tutorial examples.
>
> Compiled APIs:
>  RtAudio Dummy
>
> RtApiDummy: This class provides no functionality.
>
>
> Current API: RtAudio Dummy
>
> Found 0 device(s) ...
>
>
> I need to read wav file and record some spoken command to read by
> RtAudio. But I could not proceed.
>
> Please help me and give me some hints to proceed.
>
> Thanks in advance,
> Jui
>