添加一些关于midi播放的项目
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terryLP
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/******************************************/
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/*
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testall.cpp
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by Gary P. Scavone, 2007-2008
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This program will make a variety of calls
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to extensively test RtAudio functionality.
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*/
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/******************************************/
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#include "RtAudio.h"
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#include <iostream>
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#include <cstdlib>
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#include <cstring>
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#define BASE_RATE 0.005
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#define TIME 1.0
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void usage( void ) {
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// Error function in case of incorrect command-line
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// argument specifications
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std::cout << "\nuseage: testall N fs <iDevice> <oDevice> <iChannelOffset> <oChannelOffset>\n";
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std::cout << " where N = number of channels,\n";
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std::cout << " fs = the sample rate,\n";
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std::cout << " iDevice = optional input device to use (default = 0),\n";
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std::cout << " oDevice = optional output device to use (default = 0),\n";
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std::cout << " iChannelOffset = an optional input channel offset (default = 0),\n";
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std::cout << " and oChannelOffset = optional output channel offset (default = 0).\n\n";
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exit( 0 );
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}
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unsigned int channels;
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// Interleaved buffers
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int sawi( void *outputBuffer, void * /*inputBuffer*/, unsigned int nBufferFrames,
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double /*streamTime*/, RtAudioStreamStatus status, void *data )
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{
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unsigned int i, j;
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extern unsigned int channels;
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double *buffer = (double *) outputBuffer;
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double *lastValues = (double *) data;
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if ( status )
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std::cout << "Stream underflow detected!" << std::endl;
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for ( i=0; i<nBufferFrames; i++ ) {
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for ( j=0; j<channels; j++ ) {
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*buffer++ = (double) lastValues[j];
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lastValues[j] += BASE_RATE * (j+1+(j*0.1));
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if ( lastValues[j] >= 1.0 ) lastValues[j] -= 2.0;
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}
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}
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return 0;
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}
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// Non-interleaved buffers
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int sawni( void *outputBuffer, void * /*inputBuffer*/, unsigned int nBufferFrames,
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double /*streamTime*/, RtAudioStreamStatus status, void *data )
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{
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unsigned int i, j;
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extern unsigned int channels;
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double *buffer = (double *) outputBuffer;
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double *lastValues = (double *) data;
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if ( status )
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std::cout << "Stream underflow detected!" << std::endl;
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double increment;
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for ( j=0; j<channels; j++ ) {
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increment = BASE_RATE * (j+1+(j*0.1));
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for ( i=0; i<nBufferFrames; i++ ) {
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*buffer++ = (double) lastValues[j];
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lastValues[j] += increment;
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if ( lastValues[j] >= 1.0 ) lastValues[j] -= 2.0;
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}
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}
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return 0;
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}
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int inout( void *outputBuffer, void *inputBuffer, unsigned int /*nBufferFrames*/,
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double /*streamTime*/, RtAudioStreamStatus status, void *data )
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{
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// Since the number of input and output channels is equal, we can do
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// a simple buffer copy operation here.
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if ( status ) std::cout << "Stream over/underflow detected." << std::endl;
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unsigned int *bytes = (unsigned int *) data;
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memcpy( outputBuffer, inputBuffer, *bytes );
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return 0;
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}
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int main( int argc, char *argv[] )
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{
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unsigned int bufferFrames, fs, oDevice = 0, iDevice = 0, iOffset = 0, oOffset = 0;
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char input;
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// minimal command-line checking
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if (argc < 3 || argc > 7 ) usage();
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RtAudio dac;
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if ( dac.getDeviceCount() < 1 ) {
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std::cout << "\nNo audio devices found!\n";
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exit( 1 );
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}
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channels = (unsigned int) atoi( argv[1] );
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fs = (unsigned int) atoi( argv[2] );
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if ( argc > 3 )
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iDevice = (unsigned int) atoi( argv[3] );
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if ( argc > 4 )
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oDevice = (unsigned int) atoi(argv[4]);
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if ( argc > 5 )
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iOffset = (unsigned int) atoi(argv[5]);
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if ( argc > 6 )
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oOffset = (unsigned int) atoi(argv[6]);
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double *data = (double *) calloc( channels, sizeof( double ) );
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// Let RtAudio print messages to stderr.
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dac.showWarnings( true );
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// Set our stream parameters for output only.
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bufferFrames = 512;
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RtAudio::StreamParameters oParams, iParams;
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oParams.deviceId = oDevice;
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oParams.nChannels = channels;
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oParams.firstChannel = oOffset;
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if ( oDevice == 0 )
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oParams.deviceId = dac.getDefaultOutputDevice();
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RtAudio::StreamOptions options;
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options.flags = RTAUDIO_HOG_DEVICE;
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try {
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dac.openStream( &oParams, NULL, RTAUDIO_FLOAT64, fs, &bufferFrames, &sawi, (void *)data, &options );
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std::cout << "\nStream latency = " << dac.getStreamLatency() << std::endl;
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// Start the stream
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dac.startStream();
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std::cout << "\nPlaying ... press <enter> to stop.\n";
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std::cin.get( input );
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// Stop the stream
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dac.stopStream();
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// Restart again
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std::cout << "Press <enter> to restart.\n";
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std::cin.get( input );
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dac.startStream();
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// Test abort function
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std::cout << "Playing again ... press <enter> to abort.\n";
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std::cin.get( input );
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dac.abortStream();
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// Restart another time
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std::cout << "Press <enter> to restart again.\n";
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std::cin.get( input );
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dac.startStream();
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std::cout << "Playing again ... press <enter> to close the stream.\n";
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std::cin.get( input );
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}
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catch ( RtAudioError& e ) {
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e.printMessage();
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goto cleanup;
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}
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if ( dac.isStreamOpen() ) dac.closeStream();
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// Test non-interleaved functionality
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options.flags = RTAUDIO_NONINTERLEAVED;
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try {
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dac.openStream( &oParams, NULL, RTAUDIO_FLOAT64, fs, &bufferFrames, &sawni, (void *)data, &options );
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std::cout << "Press <enter> to start non-interleaved playback.\n";
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std::cin.get( input );
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// Start the stream
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dac.startStream();
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std::cout << "\nPlaying ... press <enter> to stop.\n";
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std::cin.get( input );
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}
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catch ( RtAudioError& e ) {
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e.printMessage();
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goto cleanup;
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}
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if ( dac.isStreamOpen() ) dac.closeStream();
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// Now open a duplex stream.
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unsigned int bufferBytes;
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iParams.deviceId = iDevice;
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iParams.nChannels = channels;
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iParams.firstChannel = iOffset;
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if ( iDevice == 0 )
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iParams.deviceId = dac.getDefaultInputDevice();
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options.flags = RTAUDIO_NONINTERLEAVED;
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try {
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dac.openStream( &oParams, &iParams, RTAUDIO_SINT32, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options );
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bufferBytes = bufferFrames * channels * 4;
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std::cout << "Press <enter> to start duplex operation.\n";
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std::cin.get( input );
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// Start the stream
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dac.startStream();
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std::cout << "\nRunning ... press <enter> to stop.\n";
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std::cin.get( input );
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// Stop the stream
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dac.stopStream();
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std::cout << "\nStopped ... press <enter> to restart.\n";
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std::cin.get( input );
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// Restart the stream
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dac.startStream();
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std::cout << "\nRunning ... press <enter> to stop.\n";
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std::cin.get( input );
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}
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catch ( RtAudioError& e ) {
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e.printMessage();
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}
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cleanup:
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if ( dac.isStreamOpen() ) dac.closeStream();
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free( data );
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return 0;
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}
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