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RWAudio_IO.cpp
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//////////////////////////////////////////////////////////////////////
// RWAudio_IO.cpp: impementation of audio interface
//
//////////////////////////////////////////////////////////////////////
/*
* Copyright (C) 2008 Vaclav Peroutka <[email protected]>
*
* Licensed under the GNU General Public License Version 2
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <https://www.gnu.org/licenses/>.
*/
#include <wx/defs.h> // for WXUNUSED
#include "RWAudio_IO.h"
#include <math.h>
#include <stdio.h>
#include <atomic>
#include <map>
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
// extern variables from AudMeS.cpp
extern float *g_OscBuffer_Left;
extern float *g_OscBuffer_Right;
extern long int g_OscBufferPosition;
extern float *g_SpeBuffer_Left;
extern float *g_SpeBuffer_Right;
extern long int g_SpeBufferPosition;
extern std::atomic<bool> g_OscBufferChanged;
extern std::atomic<bool> g_SpeBufferChanged;
/*
* pseudo noise generator - linear feedback shift register
* derived from https://en.wikipedia.org/wiki/Linear-feedback_shift_register
*/
bool lfsr16() {
uint16_t bit; /* Must be 16-bit to allow bit<<15 later in the code */
static uint16_t lfsr = 1; /* Must not be 0 */
/* taps: 16 14 13 11; feedback polynomial: x^16 + x^14 + x^13 + x^11 + 1 */
bit = ((lfsr >> 0) ^ (lfsr >> 2) ^ (lfsr >> 3) ^ (lfsr >> 5)) & 1u;
lfsr = (lfsr >> 1) | (bit << 15);
return bit;
}
/*
* callback function to catch runtime errors
*/
void catcherr(RtAudioError::Type WXUNUSED(type), const std::string &errorText) {
std::cerr << '\n' << errorText << '\n' << std::endl;
}
/*
* callback function to fetch audio input and generate tones
*/
int inout(void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
double WXUNUSED(streamTime), RtAudioStreamStatus status, void *data) {
RWAudio *aRWAudioClass = (RWAudio *)data;
unsigned i;
if (status) std::cerr << "Audio stream over/underflow detected." << std::endl;
// copy input buffer into two L/R channels
if (aRWAudioClass->m_Buflen_Changed) {
aRWAudioClass->m_Buflen_Changed = false;
free(g_OscBuffer_Left);
free(g_OscBuffer_Right);
free(g_SpeBuffer_Left);
free(g_SpeBuffer_Right);
g_OscBufferPosition = 0;
g_SpeBufferPosition = 0;
g_OscBuffer_Left = (float *)malloc(aRWAudioClass->m_OscBufferLen * sizeof(float));
g_OscBuffer_Right = (float *)malloc(aRWAudioClass->m_OscBufferLen * sizeof(float));
g_SpeBuffer_Left = (float *)malloc(aRWAudioClass->m_SpeBufferLen * sizeof(float));
g_SpeBuffer_Right = (float *)malloc(aRWAudioClass->m_SpeBufferLen * sizeof(float));
}
float *inBuf = (float *)inputBuffer;
// make a copy for oscilloscope
if (!g_OscBufferChanged.load()) {
for (i = 0; i < nBufferFrames; i++) {
// copy audio signal to fft real component.
g_OscBuffer_Left[g_OscBufferPosition] = *inBuf++;
if (aRWAudioClass->m_channels_in > 1)
g_OscBuffer_Right[g_OscBufferPosition] = *inBuf++;
else
g_OscBuffer_Right[g_OscBufferPosition] = 0;
g_OscBufferPosition++;
// if the buffer is over we have to pick the data and then circullary fill the next one
if (g_OscBufferPosition == aRWAudioClass->m_OscBufferLen) {
g_OscBufferPosition = 0;
g_OscBufferChanged = true;
break;
}
}
}
inBuf = (float *)inputBuffer;
// make a copy for spectrum analyzer
if (!g_SpeBufferChanged.load()) {
for (i = 0; i < nBufferFrames; i++) {
// copy audio signal to fft real component.
g_SpeBuffer_Left[g_SpeBufferPosition] = *inBuf++;
if (aRWAudioClass->m_channels_in > 1)
g_SpeBuffer_Right[g_SpeBufferPosition] = *inBuf++;
else
g_SpeBuffer_Right[g_SpeBufferPosition] = 0;
g_SpeBufferPosition++;
// if the buffer is over we have to pick the data and then circullary fill the next one
if (g_SpeBufferPosition == aRWAudioClass->m_SpeBufferLen) {
g_SpeBufferPosition = 0;
g_SpeBufferChanged = true;
break;
}
}
}
// fill output buffer
float *outBuf = (float *)outputBuffer;
#ifdef _DEBUG
// fprintf(ddbg, "\n Frames: %d\n ", nBufferFrames);
#endif
/* calculate the wave form according to the selected shape */
for (i = 0; i < nBufferFrames; i++) {
double y = 0;
double y2 = 0;
bool noise = lfsr16();
/* left channel */
switch (aRWAudioClass->m_genShape_l) {
case 1:
if (aRWAudioClass->m_genPhase_l < M_PI) {
y = 1.0;
} else {
y = -1.0;
}
break;
case 2:
y = (aRWAudioClass->m_genPhase_l - M_PI) / M_PI;
break;
case 3:
if (aRWAudioClass->m_genPhase_l < M_PI) {
y = 2 * (aRWAudioClass->m_genPhase_l - M_PI / 2) / M_PI;
} else {
y = 2 * (3 * M_PI / 2 - aRWAudioClass->m_genPhase_l) / M_PI;
}
break;
case 4:
if (noise) {
y = 1.0;
} else {
y = -1.0;
}
break;
default: /* sine wave */
y = sin(aRWAudioClass->m_genPhase_l);
break;
}
/* right channel */
switch (aRWAudioClass->m_genShape_r) {
case 1:
if ((aRWAudioClass->m_genPhase_r - aRWAudioClass->m_genPhaseDif) < M_PI) {
y2 = 1;
} else {
y2 = -1;
}
break;
case 2:
y2 = ((aRWAudioClass->m_genPhase_r - aRWAudioClass->m_genPhaseDif) - M_PI) / M_PI;
break;
case 3:
if (aRWAudioClass->m_genPhase_r < M_PI) {
y2 = 2 * (aRWAudioClass->m_genPhase_r - aRWAudioClass->m_genPhaseDif - M_PI / 2) / M_PI;
} else {
y2 = 2 * (3 * M_PI / 2 - aRWAudioClass->m_genPhase_r + aRWAudioClass->m_genPhaseDif) /
M_PI;
}
break;
case 4:
if (noise) {
y2 = 1.0;
} else {
y2 = -1.0;
}
break;
default: /* sine wave */
y2 = sin(aRWAudioClass->m_genPhase_r - aRWAudioClass->m_genPhaseDif);
break;
}
if (aRWAudioClass->m_genGain_l == 0.0)
aRWAudioClass->m_genPhase_l = 0.0;
else
aRWAudioClass->m_genPhase_l +=
(float)2.0 * M_PI * aRWAudioClass->m_genFR_l / aRWAudioClass->m_sampleRate;
if (aRWAudioClass->m_genGain_r == 0.0)
aRWAudioClass->m_genPhase_r = 0.0;
else
aRWAudioClass->m_genPhase_r +=
(float)2.0 * M_PI * aRWAudioClass->m_genFR_r / aRWAudioClass->m_sampleRate;
if ((2.0 * M_PI) < aRWAudioClass->m_genPhase_l) aRWAudioClass->m_genPhase_l -= 2.0 * M_PI;
if ((2.0 * M_PI) < aRWAudioClass->m_genPhase_r) aRWAudioClass->m_genPhase_r -= 2.0 * M_PI;
*outBuf++ = (float)(aRWAudioClass->m_genGain_l * y);
if (aRWAudioClass->m_channels_out > 1) *outBuf++ = (float)(aRWAudioClass->m_genGain_r * y2);
#ifdef _DEBUG
// fprintf(ddbg,"%04X %04X ",(float)(32768.f * y), (float)(32768.f * y2));
#endif
}
return 0;
}
RWAudio::RWAudio() {
m_Buflen_Changed = false;
m_sampleRate = 0;
}
RWAudio::~RWAudio() {
m_AudioDriver->stopStream();
m_AudioDriver->closeStream();
}
// Initialize RtAudio and start audio stream
int RWAudio::InitSnd(long int oscbuflen, long int spebuflen, std::string &rtinfo,
unsigned int srate) {
m_OscBufferLen = oscbuflen;
m_SpeBufferLen = spebuflen;
m_sampleRate = srate;
m_genFR_l = m_genFR_r = 0.0;
m_genShape_l = m_genShape_r = 0;
m_genGain_l = m_genGain_r = 1.0;
m_genPhase_l = m_genPhase_r = 0.0;
m_genPhaseDif = 0.0;
g_OscBufferPosition = 0;
g_SpeBufferPosition = 0;
g_OscBuffer_Left = (float *)malloc(m_OscBufferLen * sizeof(float));
g_OscBuffer_Right = (float *)malloc(m_OscBufferLen * sizeof(float));
g_SpeBuffer_Left = (float *)malloc(m_SpeBufferLen * sizeof(float));
g_SpeBuffer_Right = (float *)malloc(m_SpeBufferLen * sizeof(float));
RtAudio::DeviceInfo info;
unsigned int devices;
// Create an api map.
std::map<int, std::string> apiMap;
apiMap[RtAudio::MACOSX_CORE] = "OS-X Core Audio";
apiMap[RtAudio::WINDOWS_ASIO] = "Windows ASIO";
apiMap[RtAudio::WINDOWS_DS] = "Windows DirectSound";
apiMap[RtAudio::WINDOWS_WASAPI] = "Windows WASAPI";
apiMap[RtAudio::UNIX_JACK] = "Jack Client";
apiMap[RtAudio::LINUX_ALSA] = "Linux ALSA";
apiMap[RtAudio::LINUX_PULSE] = "Linux PulseAudio";
apiMap[RtAudio::LINUX_OSS] = "Linux OSS";
apiMap[RtAudio::RTAUDIO_DUMMY] = "RtAudio Dummy";
std::vector<RtAudio::Api> apis;
// init of RtAudio
RtAudio::getCompiledApi(apis);
rtinfo = "\nRtAudio Version ";
rtinfo = rtinfo + RtAudio::getVersion() + "\nCompiled APIs:\n";
for (unsigned int i = 0; i < apis.size(); i++) rtinfo = rtinfo + " " + apiMap[apis[i]] + "\n";
m_AudioDriver = new RtAudio();
rtinfo = rtinfo + "Current API: " + apiMap[m_AudioDriver->getCurrentApi()] + "\n";
devices = m_AudioDriver->getDeviceCount();
if (devices < 1) return 1;
for (unsigned int i = 0; i < devices; i++) {
info = m_AudioDriver->getDeviceInfo(i);
}
// start audio streams
if (RestartAudio(m_AudioDriver->getDefaultInputDevice(),
m_AudioDriver->getDefaultOutputDevice())) {
return 2;
}
return 0;
}
int RWAudio::RestartAudio(int recDevId, int playDevId) {
// if stream is open (and running), stop it
if (m_AudioDriver->isStreamOpen()) {
m_AudioDriver->stopStream();
m_AudioDriver->closeStream();
}
// adapt to number of channels
RtAudio::DeviceInfo info = m_AudioDriver->getDeviceInfo(recDevId);
if (info.inputChannels > 1 || info.duplexChannels > 1)
m_channels_in = 2;
else
m_channels_in = 1;
info = m_AudioDriver->getDeviceInfo(playDevId);
if (info.outputChannels > 1 || info.duplexChannels > 1)
m_channels_out = 2;
else
m_channels_out = 1;
// configure new stream
RtAudio::StreamParameters iParams;
RtAudio::StreamParameters oParams;
RtAudio::StreamOptions rtAOptions;
unsigned int bufferFrames = 512;
iParams.deviceId = recDevId;
oParams.deviceId = playDevId;
iParams.nChannels = m_channels_in;
oParams.nChannels = m_channels_out;
iParams.firstChannel = 0;
oParams.firstChannel = 0;
rtAOptions.flags = 0;
try {
m_AudioDriver->openStream(&oParams, &iParams, RTAUDIO_FLOAT32, m_sampleRate, &bufferFrames,
&inout, (void *)this, &rtAOptions, &catcherr);
} catch (RtAudioError &e) {
// std::cerr << '\n' << e.getMessage() << '\n' << std::endl;
return 1;
}
try {
m_AudioDriver->startStream();
} catch (RtAudioError &e) {
// std::cerr << '\n' << e.getMessage() << '\n' << std::endl;
return 1;
}
return 0;
}
/********************************************************************/
/************* Devices enumeration *******************/
/********************************************************************/
int RWAudio::GetRWAudioDevices(RWAudioDevList *play, RWAudioDevList *record) {
// Determine the number of devices available
unsigned int devices = m_AudioDriver->getDeviceCount();
// Scan through devices for various capabilities
RtAudio::DeviceInfo info;
// if stream is open (and running), stop it
if (m_AudioDriver->isStreamOpen()) {
m_AudioDriver->stopStream();
m_AudioDriver->closeStream();
}
play->card_info.clear();
record->card_info.clear();
play->card_pos.clear();
record->card_pos.clear();
for (unsigned int i = 0; i < devices; i++) {
info = m_AudioDriver->getDeviceInfo(i);
if (info.probed == true) {
// std::cout << "device = " << i << "; name: " << info.name << "\n";
// add play card
if ((info.outputChannels > 0) || (info.duplexChannels > 0)) {
play->card_info.push_back(info);
play->card_pos.push_back(i);
}
// add record card
if ((info.inputChannels > 0) || (info.duplexChannels > 0)) {
record->card_info.push_back(info);
record->card_pos.push_back(i);
}
}
}
return 0;
}
/********************************************************************/
/************* Parameter settings *******************/
/********************************************************************/
int RWAudio::PlaySetGenerator(float f1, float f2, int s1, int s2, float g1, float g2) {
if (2 * f1 < m_sampleRate)
m_genFR_l = f1;
else
m_genFR_l = m_sampleRate / 2;
if (2 * f2 < m_sampleRate)
m_genFR_r = f2;
else
m_genFR_r = m_sampleRate / 2;
m_genShape_l = s1;
m_genShape_r = s2;
if (1.0 > g1) {
m_genGain_l = g1;
} else {
m_genGain_l = 0.99999;
}
if (1.0 > g2) {
m_genGain_r = g2;
} else {
m_genGain_r = 0.99999;
}
return 1;
}
void RWAudio::SetSndDevices(unsigned int irec, unsigned int iplay, unsigned int srate) {
unsigned int cardrec, cardplay;
cardrec = irec;
cardplay = iplay;
m_sampleRate = srate;
// redefine audio streams
RestartAudio(cardrec, cardplay);
}