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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playsaw.cpp
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by Gary P. Scavone, 2006
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This program will output sawtooth waveforms
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of different frequencies on each channel.
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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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#include <map>
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#include "Player.h"
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/*
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typedef char MY_TYPE;
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#define FORMAT RTAUDIO_SINT8
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#define SCALE 127.0
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*/
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typedef signed short MY_TYPE;
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#define FORMAT RTAUDIO_SINT16
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#define SCALE 32767.0
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/*
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typedef S24 MY_TYPE;
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#define FORMAT RTAUDIO_SINT24
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#define SCALE 8388607.0
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typedef signed long MY_TYPE;
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#define FORMAT RTAUDIO_SINT32
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#define SCALE 2147483647.0
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typedef float MY_TYPE;
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#define FORMAT RTAUDIO_FLOAT32
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#define SCALE 1.0
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typedef double MY_TYPE;
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#define FORMAT RTAUDIO_FLOAT64
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#define SCALE 1.0
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*/
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// Platform-dependent sleep routines.
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#if defined( __WINDOWS_ASIO__ ) || defined( __WINDOWS_DS__ ) || defined( __WINDOWS_WASAPI__ )
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#include <windows.h>
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#define SLEEP( milliseconds ) Sleep( (DWORD) milliseconds )
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#else // Unix variants
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#include <unistd.h>
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#define SLEEP( milliseconds ) usleep( (unsigned long) (milliseconds * 1000.0) )
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#endif
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#define BASE_RATE 0.005
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#define TIME 1.0
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void errorCallback(RtAudioError::Type type, const std::string &errorText)
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{
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// This example error handling function does exactly the same thing
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// as the embedded RtAudio::error() function.
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std::cout << "in errorCallback" << std::endl;
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if (type == RtAudioError::WARNING)
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std::cerr << '\n' << errorText << "\n\n";
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else if (type != RtAudioError::WARNING)
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throw(RtAudioError(errorText, type));
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}
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unsigned int channels;
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RtAudio::StreamOptions options;
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unsigned int frameCounter = 0;
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bool checkCount = false;
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unsigned int nFrames = 0;
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const unsigned int callbackReturnValue = 1;
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void ShowAudioDevice(void)
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{
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// Create an api map.
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std::map<int, std::string> apiMap;
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apiMap[RtAudio::MACOSX_CORE] = "OS-X Core Audio";
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apiMap[RtAudio::WINDOWS_ASIO] = "Windows ASIO";
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apiMap[RtAudio::WINDOWS_DS] = "Windows Direct Sound";
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apiMap[RtAudio::WINDOWS_WASAPI] = "Windows WASAPI";
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apiMap[RtAudio::UNIX_JACK] = "Jack Client";
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apiMap[RtAudio::LINUX_ALSA] = "Linux ALSA";
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apiMap[RtAudio::LINUX_PULSE] = "Linux PulseAudio";
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apiMap[RtAudio::LINUX_OSS] = "Linux OSS";
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apiMap[RtAudio::RTAUDIO_DUMMY] = "RtAudio Dummy";
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std::vector< RtAudio::Api > apis;
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RtAudio::getCompiledApi(apis);
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std::cout << "\nRtAudio Version " << RtAudio::getVersion() << std::endl;
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std::cout << "\nCompiled APIs:\n";
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for (unsigned int i = 0; i < apis.size(); i++)
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std::cout << " " << apiMap[apis[i]] << std::endl;
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RtAudio audio;
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RtAudio::DeviceInfo info;
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std::cout << "\nCurrent API: " << apiMap[audio.getCurrentApi()] << std::endl;
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unsigned int devices = audio.getDeviceCount();
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std::cout << "\nFound " << devices << " device(s) ...\n";
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for (unsigned int i = 0; i < devices; i++) {
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info = audio.getDeviceInfo(i);
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std::cout << "\nDevice Name = " << info.name << '\n';
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std::cout << "Device ID = " << i << '\n';
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if (info.probed == false)
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std::cout << "Probe Status = UNsuccessful\n";
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else {
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std::cout << "Probe Status = Successful\n";
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std::cout << "Output Channels = " << info.outputChannels << '\n';
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std::cout << "Input Channels = " << info.inputChannels << '\n';
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std::cout << "Duplex Channels = " << info.duplexChannels << '\n';
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if (info.isDefaultOutput) std::cout << "This is the default output device.\n";
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else std::cout << "This is NOT the default output device.\n";
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if (info.isDefaultInput) std::cout << "This is the default input device.\n";
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else std::cout << "This is NOT the default input device.\n";
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if (info.nativeFormats == 0)
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std::cout << "No natively supported data formats(?)!";
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else {
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std::cout << "Natively supported data formats:\n";
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if (info.nativeFormats & RTAUDIO_SINT8)
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std::cout << " 8-bit int\n";
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if (info.nativeFormats & RTAUDIO_SINT16)
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std::cout << " 16-bit int\n";
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if (info.nativeFormats & RTAUDIO_SINT24)
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std::cout << " 24-bit int\n";
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if (info.nativeFormats & RTAUDIO_SINT32)
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std::cout << " 32-bit int\n";
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if (info.nativeFormats & RTAUDIO_FLOAT32)
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std::cout << " 32-bit float\n";
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if (info.nativeFormats & RTAUDIO_FLOAT64)
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std::cout << " 64-bit float\n";
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}
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if (info.sampleRates.size() < 1)
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std::cout << "No supported sample rates found!";
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else {
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std::cout << "Supported sample rates = ";
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for (unsigned int j = 0; j < info.sampleRates.size(); j++)
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std::cout << info.sampleRates[j] << " ";
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}
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std::cout << std::endl;
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}
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}
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}
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//#define USE_INTERLEAVED
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#if defined( USE_INTERLEAVED )
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// Interleaved buffers
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int audioRenderCallback(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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MY_TYPE *buffer = (MY_TYPE *)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++ = (MY_TYPE)(lastValues[j] * SCALE * 0.5);
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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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frameCounter += nBufferFrames;
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if (checkCount && (frameCounter >= nFrames)) return callbackReturnValue;
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return 0;
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}
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#else // Use non-interleaved buffers
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int audioRenderCallback(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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MY_TYPE *buffer = (MY_TYPE *)outputBuffer;
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Player *pPlayer = (Player *)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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Player32kProc(pPlayer);
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int32_t rawSynthOutput = pPlayer->mainSynthesizer.mixOut >> 10;
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if (rawSynthOutput < -32768)
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rawSynthOutput = -32768;
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else if (rawSynthOutput > 32767)
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rawSynthOutput = 32767;
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*buffer++ = (MY_TYPE)rawSynthOutput;
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}
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memcpy(buffer, outputBuffer, nBufferFrames * sizeof(MY_TYPE));
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PlayerProcess(pPlayer);
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return 0;
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}
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#endif
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int main(int argc, char *argv[])
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{
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unsigned int bufferFrames, fs, device = 0, offset = 0;
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ShowAudioDevice();
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RtAudio dac;
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Player player;
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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 = 2;
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fs = 32000;
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device = 0;
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PlayerInit(&player);
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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 = 64;
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RtAudio::StreamParameters oParams;
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oParams.deviceId = device;
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oParams.nChannels = channels;
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oParams.firstChannel = offset;
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if (device == 0)
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{
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oParams.deviceId = dac.getDefaultOutputDevice();
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std::cout << "Select default output device: " << oParams.deviceId << '\n';
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}
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options.flags = RTAUDIO_HOG_DEVICE;
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options.flags |= RTAUDIO_SCHEDULE_REALTIME;
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#if !defined( USE_INTERLEAVED )
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options.flags |= RTAUDIO_NONINTERLEAVED;
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#endif
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try {
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dac.openStream(&oParams, NULL, FORMAT, fs, &bufferFrames, &audioRenderCallback, (void *)&player, &options, &errorCallback);
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dac.startStream();
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PlayerPlay(&player);
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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 (checkCount) {
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while (dac.isStreamRunning() == true) SLEEP(100);
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}
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else {
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char input;
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//std::cout << "Stream latency = " << dac.getStreamLatency() << "\n" << std::endl;
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std::cout << "\nPlaying ... press <enter> to quit (buffer size = " << bufferFrames << ").\n";
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std::cin.get(input);
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try {
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// Stop the stream
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dac.stopStream();
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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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}
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cleanup:
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if (dac.isStreamOpen()) dac.closeStream();
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return 0;
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}
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