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platform.c++
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platform.c++
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// platform-specific code
#ifdef VSS_WINDOWS
#include "windows.h"
#include "winplatform.h"
#include <sys/select.h>
#endif
#ifdef VSS_IRIX
#include <dmedia/audio.h>
#include <stropts.h>
#include <sys/lock.h>
#include <sys/prctl.h>
#include <sys/schedctl.h>
#if defined VSS_IRIX_62 || defined VSS_IRIX_53
#define alGetFD(_) ALgetfd(_)
#define alNewConfig() ALnewconfig()
#define alSetWidth(_,__) ALsetwidth(_,__)
#define alSetQueueSize(_,__) ALsetqueuesize(_,__)
#define alSetChannels(_,__) ALsetchannels(_,__)
#define alOpenPort(_,__,___) ALopenport(_,__,___)
#define alSetParams(_,__,___) ALsetparams(_,__,___) /* not quite right! */
#define alGetFD(_) ALgetfd(_)
#define alGetFilled(_) ALgetfilled(_)
#define alClosePort(_) ALcloseport(_)
#define alReadFrames(_,__,___) ALreadsamps(_,__,___*nchansIn) // in Synth()
#define alWriteFrames(_,__,___) ALwritesamps(_,__,___*nchans) // in Synth()
#define alSetFillPoint(_,__) ALsetfillpoint(_,__*nchans) // in Synth()
#endif
#endif
#include <algorithm>
#include <arpa/inet.h>
#include <cerrno>
#include <climits>
#include <cmath>
#include <csignal>
#include <cstdio>
#include <cstdlib>
#include <cstring>
#include <ctime>
#include <fcntl.h>
#include <grp.h>
#include <iostream>
#include <netdb.h>
#include <netinet/in.h>
#include <poll.h> // struct pollfd
#include <pwd.h>
#include <sys/fcntl.h>
#include <sys/file.h>
#include <sys/ioctl.h>
#include <sys/param.h>
#include <sys/socket.h>
#include <sys/stat.h>
#include <sys/times.h>
#include <sys/types.h>
#include <unistd.h>
#include "platform.h"
#include "vssglobals.h"
#include "VAlgorithm.h" // only for dBFromScalar()
using std::cerr;
#ifdef VSS_LINUX_UBUNTU
#include <alsa/asoundlib.h> // apt install libasound2-dev
snd_pcm_t* pcm_handle_read = nullptr;
snd_pcm_t* pcm_handle_write = nullptr;
// Set by set_hwparams(). Used by set_swparams().
snd_pcm_sframes_t buffer_size;
snd_pcm_sframes_t period_size;
int xrun_recovery(snd_pcm_t* handle, int err)
{
if (err == -EPIPE) {
LPrepare:
err = snd_pcm_prepare(handle);
if (err < 0)
printf("Failed to recover from underrun: %s\n", snd_strerror(err));
return 0;
}
if (err == -ESTRPIPE) {
// Busywait until the suspend flag is released.
while ((err = snd_pcm_resume(handle)) == -EAGAIN)
usleep(100000);
if (err < 0)
goto LPrepare;
return 0;
}
return err;
}
int set_hwparams(snd_pcm_t* handle, snd_pcm_hw_params_t* params, int nchans)
{
int rc;
if ((rc = snd_pcm_hw_params_any(handle, params)) < 0) {
cerr << "No configurations for playback: " << snd_strerror(rc) << "\n";
return rc;
}
if ((rc = snd_pcm_hw_params_set_rate_resample(handle, params, 1)) < 0) {
cerr << "Hardware resampling failed for playback: " << snd_strerror(rc) << "\n";
return rc;
}
// set the interleaved read/write format
if ((rc = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
cerr << "Access type unavailable for playback: " << snd_strerror(rc) << "\n";
return rc;
}
if ((rc = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE)) < 0) {
cerr << "Sample format not available for playback: " << snd_strerror(rc) << "\n";
return rc;
}
if ((rc = snd_pcm_hw_params_set_channels(handle, params, nchans)) < 0) {
cerr << "Failed to play " << nchans << " channels: " << snd_strerror(rc) << "\n";
return rc;
}
const unsigned int rate = globs.SampleRate;
unsigned int rrate = rate;
if ((rc = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0)) < 0) {
cerr << "Failed to play at " << rate << " Hz: " << snd_strerror(rc) << "\n";
return rc;
}
if (rrate != rate) {
cerr << "Requested rate " << rate << " Hz != actual rate " << rrate << " Hz\n";
return -EINVAL;
}
auto dir = 1;
auto usec_buffer = 10000u;
if ((rc = snd_pcm_hw_params_set_buffer_time_near(handle, params, &usec_buffer, &dir)) < 0) {
cerr << "Failed to play with ring buffer of " << usec_buffer << " usec: " << snd_strerror(rc) << "\n";
return rc;
}
snd_pcm_uframes_t size;
if ((rc = snd_pcm_hw_params_get_buffer_size(params, &size)) < 0) {
cerr << "Failed to get playback buffer size: " << snd_strerror(rc) << "\n";
return rc;
}
buffer_size = size;
auto usec_period = 3000u; // As low as 500 might work, or even 32.
if ((rc = snd_pcm_hw_params_set_period_time_near(handle, params, &usec_period, &dir)) < 0) {
cerr << "Failed to play at period of " << usec_period << " usec: " << snd_strerror(rc) << "\n";
return rc;
}
if ((rc = snd_pcm_hw_params_get_period_size(params, &size, &dir)) < 0) {
cerr << "Failed to get playback period size: " << snd_strerror(rc) << "\n";
return rc;
}
period_size = size;
if ((rc = snd_pcm_hw_params(handle, params)) < 0) {
cerr << "Failed to set playback hw params: " << snd_strerror(rc) << "\n";
return rc;
}
//printf("\nSR = %d Hz.\n%d channels.\nLatency = %.1f ms.\n\n", rate, nchans, float(period_size)/rate * 1000.0);
return 0;
}
int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams)
{
int rc;
if ((rc = snd_pcm_sw_params_current(handle, swparams)) < 0) {
cerr << "Failed to get playback swparams: " << snd_strerror(rc) << "\n";
return rc;
}
/* start the transfer when the buffer is almost full: */
/* (buffer_size / avail_min) * avail_min */
if ((rc = snd_pcm_sw_params_set_start_threshold(handle, swparams, (buffer_size / period_size) * period_size)) < 0) {
cerr << "Failed to set playback start threshold: " << snd_strerror(rc) << "\n";
return rc;
}
/* allow the transfer when at least period_size samples can be processed */
if ((rc = snd_pcm_sw_params_set_avail_min(handle, swparams, period_size)) < 0) {
cerr << "Failed to set playback available min: " << snd_strerror(rc) << "\n";
return rc;
}
if ((rc = snd_pcm_sw_params(handle, swparams)) < 0) {
cerr << "Failed to set playback sw params: " << snd_strerror(rc) << "\n";
return rc;
}
return 0;
}
#endif // VSS_LINUX_UBUNTU
#ifdef VSS_LINUX
#include <linux/soundcard.h>
#include <csignal>
#ifndef VSS_LINUX_UBUNTU
int fdDAC = -1; // hardware output (and input, actually)
#endif
#elif defined VSS_IRIX
// Yucky globals.
static ALport alp;
static ALport alpin;
static ALconfig alc;
static unsigned long qsize;
static int latency;
#elif defined VSS_WINDOWS
#include "fmod.h"
#include "fmod_errors.h"
static int (*pfn)() = NULL;
#define TESTIT \
if (!pfn) { \
printf("failed to find function in fmod.dll\n"); \
FreeLibrary(fmod_dll); \
exit(1); }
static int (*_FSOUND_Stream_GetTime_hack)(FSOUND_STREAM*) = NULL;
static FSOUND_STREAM *fsound_stream = NULL;
static HMODULE fmod_dll = NULL;
static int vfCalledback = 0;
static int vfLiveTickPaused = 1; // first streamcallback happens before first LiveTick.
extern short* vrgsCallback;
short* vrgsCallback = NULL;
int vcbCallback = 0;
extern int vfMMIO;
int vfMMIO = 0;
#endif // VSS_WINDOWS
static auto vfDie = false; // Set to true when vss is dying.
int vfSoftClip = false;
int vfLimitClip = false;
constexpr auto wSoftclipLim = 50000; // Start clipping at +-25000, about -3 dB.
static int rgwSoftclip[wSoftclipLim + 1] = {0};
constexpr auto NSAMPS = MaxSampsPerBuffer * MaxNumChannels; /* or even more! */
static short sampbuff[NSAMPS] = {0};
static short* ssp; // into sampbuff
static float outvecp[NSAMPS] = {0};
static float inpvecp[NSAMPS] = {0};
static short ibuf [NSAMPS] = {0};
static bool fSoundIn = false;
extern void SetSoundIn(int f) { fSoundIn = f; } // Only misc.c++.
const float* VssInputBuffer() { return fSoundIn ? inpvecp : nullptr; }
static bool vfWaitForReinit = false;
int VSSglobals::Initsynth() {
vfWaitForReinit = true;
usleep(250000); // make sure that LiveTick() noticed that we set vfWaitForReinit, and is now waiting.
nchansOut = std::clamp(nchansOut, 1, MaxNumChannels);
nchansIn = std::clamp(nchansIn, 1, MaxNumChannels);
ssp = sampbuff;
fdOfile = -1;
vcbBufOfile = 0;
vibBufOfile = 0;
rgbBufOfile = nullptr;
// liveaudio is a badly-named flag indicating that
// samples are being scheduled and sent to a CODEC
// in real time, rather than being dumped to a file.
if (!liveaudio)
{
// ParseArgs() already checked this.
nchansIn = 0;
fSoundIn = false;
}
else
{
#ifdef VSS_IRIX
alc = alNewConfig();
alSetWidth(alc, AL_SAMPLE_16);
latency = lat;
qsize = hwm;
alSetQueueSize (alc, (int)qsize);
if (alSetChannels(alc, (long)nchansOut) < 0)
{
cerr << "vss: couldn't play " << nchansOut << " channels, using 1 instead.\n";
nchansOut = 1;
}
alp = alOpenPort("obuf", "w", alc);
if (!alp)
{
#ifdef VSS_IRIX_63PLUS
cerr << "vss: failed to output audio: " << alGetErrorString(oserror());
#endif
return -1; /* no audio hardware */
}
if (fSoundIn)
{
if (alSetChannels(alc, nchansIn) < 0)
{
cerr << "vss: couldn't input " << nchansIn << " channels, using 1 instead.\n";
nchansIn = 1;
alSetChannels(alc, nchansIn);
}
alpin = alOpenPort("ibuf", "r", alc);
if (!alpin)
{
#ifdef VSS_IRIX_63PLUS
cerr << "vss: failed to input audio: " << alGetErrorString(oserror());
#endif
nchansIn = 0;
}
}
#if defined VSS_IRIX_62 || defined VSS_IRIX_53
long pvbuf[6];
long pvlen;
pvbuf[0] = AL_OUTPUT_RATE;
pvbuf[1] = AL_RATE_44100;
pvlen = 2;
if(srate== 8000) pvbuf[1] = AL_RATE_8000;
else if(srate== 11025) pvbuf[1] = AL_RATE_11025;
else if(srate== 16000) pvbuf[1] = AL_RATE_16000;
else if(srate== 22050) pvbuf[1] = AL_RATE_22050;
else if(srate== 32000) pvbuf[1] = AL_RATE_32000;
else if(srate== 44100) pvbuf[1] = AL_RATE_44100;
else if(srate== 48000) pvbuf[1] = AL_RATE_48000;
else
fprintf(stderr, "unsupported sample rate %f, using %d instead\n",srate,44100);
if (fSoundIn)
{
pvbuf[pvlen] = AL_INPUT_RATE;
pvbuf[pvlen+1] = pvbuf[1];
ALsetparams(AL_DEFAULT_DEVICE, pvbuf, pvlen);
pvlen += 2;
}
if (nchansOut == 4)
{
pvbuf[pvlen] = AL_CHANNEL_MODE;
pvbuf[pvlen+1] = 4;
pvlen += 2;
}
ALsetparams(AL_DEFAULT_DEVICE, pvbuf, pvlen);
#else // VSS_IRIX_63 or VSS_IRIX_65
ALpv pvbuf[3];
long npvs = 0;
pvbuf[npvs].param = AL_RATE;
pvbuf[npvs].value.ll = alDoubleToFixed(srate);
++npvs;
if (nchansOut == 4)
{
pvbuf[npvs].param = AL_CHANNEL_MODE;
pvbuf[npvs].value.i = 4;
++npvs;
}
if (alSetParams(AL_DEFAULT_OUTPUT, pvbuf, npvs) < 0)
{
cerr << "vss: alSetParams failed: " << alGetErrorString(oserror()) << "\n";
if (pvbuf[1].sizeOut < 0)
cerr << "vss: invalid output sample rate " << srate << ".\n";
}
npvs = 0;
if (fSoundIn)
{
pvbuf[npvs].param = AL_RATE;
pvbuf[npvs].value = pvbuf[0].value;
++npvs;
if (alSetParams(AL_DEFAULT_INPUT, pvbuf, npvs) < 0)
{
cerr << "vss: alSetParams failed: " << alGetErrorString(oserror()) << "\n";
if (pvbuf[1].sizeOut < 0)
cerr << "vss: invalid input sample rate " << srate << ".\n";
}
}
#endif // VSS_IRIX_63 or VSS_IRIX_65
#endif // VSS_IRIX
#ifdef VSS_LINUX_UBUNTU
int rc;
if ((rc = snd_pcm_open(&pcm_handle_write, "default" /* or e.g. "hw:0,0" */, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
// ALSA prints a dozen errors first, even without snd_output_stdio_attach().
cerr << "vss: no audio out: " << snd_strerror(rc) << "\n";
liveaudio = 0;
goto LContinue;
}
snd_pcm_hw_params_t* hwparams;
snd_pcm_hw_params_alloca(&hwparams);
if ((rc = set_hwparams(pcm_handle_write, hwparams, NchansOut())) < 0) {
// set_hwparams already complained.
liveaudio = 0;
goto LContinue;
}
snd_pcm_sw_params_t* swparams;
snd_pcm_sw_params_alloca(&swparams);
if ((rc = set_swparams(pcm_handle_write, swparams)) < 0) {
cerr << "vss failed to set swparams: " << snd_strerror(rc) << "\n";
liveaudio = 0;
goto LContinue;
}
if (fSoundIn) {
if ((rc = snd_pcm_open(&pcm_handle_read, "default", SND_PCM_STREAM_CAPTURE, 0)) < 0) {
cerr << "vss: no audio in: " << snd_strerror(rc) << "\n";
liveaudio = 0;
goto LContinue;
}
if ((rc = set_hwparams(pcm_handle_read, hwparams, nchansIn)) < 0) {
// set_hwparams already complained.
fSoundIn = false;
}
if ((rc = set_swparams(pcm_handle_read, swparams)) < 0) {
cerr << "vss failed to set input swparams: " << snd_strerror(rc) << "\n";
fSoundIn = false;
}
}
#if 0
// These segfault. Maybe only after snd_pcm_close(), in CloseSynth()?
snd_pcm_hw_params_free(hwparams);
snd_pcm_sw_params_free(swparams);
#endif
#endif // VSS_LINUX_UBUNTU
#ifdef VSS_WINDOWS
if (vfMMIO)
{
if (!areal_internal_FInitAudio(nchansOut, nchansIn, fSoundIn, nchansOut*MaxSampsPerBuffer))
goto LFailed;
// Using MMIO audio i/o.
goto LContinue;
}
// Using FMOD DirectSound audio i/o.
if (!fmod_dll)
fmod_dll = LoadLibrary("fmod.dll");
if (!fmod_dll)
{
printf("vss: failed to load FMOD.DLL.\n");
goto LFailed;
}
if (nchansOut > 2)
{
printf("vss: DirectSound outputs at most 2 channels. Using 2 instead of %d.\n", nchansOut);
nchansOut = 2;
}
{
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_GetVersion@0"); TESTIT
float (*_FSOUND_GetVersion)() = (float(*)())pfn;
float version = _FSOUND_GetVersion();
if (version != FMOD_VERSION)
{
printf("vss: FMOD.DLL is version %.02f, but should be %.02f.\n", version, FMOD_VERSION);
FreeLibrary(fmod_dll);
goto LFailed;
}
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_Stream_GetTime@4"); TESTIT
_FSOUND_Stream_GetTime_hack = (int(*)(FSOUND_STREAM*))pfn;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_SetOutput@4"); TESTIT
void (*_FSOUND_SetOutput)(int) = (void(*)(int))pfn;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_SetBufferSize@4"); TESTIT
void (*_FSOUND_SetBufferSize)(int) = (void(*)(int))pfn;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_SetDriver@4"); TESTIT
void (*_FSOUND_SetDriver)(int) = (void(*)(int))pfn;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_Init@12"); TESTIT
int (*_FSOUND_Init)(int, int, int) = (int(*)(int, int, int))pfn;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_GetError@0"); TESTIT
int (*_FSOUND_GetError)() = (int(*)())pfn;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_Stream_Create@20"); TESTIT
FSOUND_STREAM* (*_FSOUND_Stream_Create)(FSOUND_STREAMCALLBACK, int, unsigned int, int, int) = (FSOUND_STREAM*(*)(FSOUND_STREAMCALLBACK, int, unsigned int, int, int))pfn;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_Stream_Play@8"); TESTIT
int (*_FSOUND_Stream_Play)(int, FSOUND_STREAM*) = (int(*)(int, FSOUND_STREAM*))pfn;
_FSOUND_SetOutput(FSOUND_OUTPUT_DSOUND);
_FSOUND_SetDriver(0); // Select sound card (0 = default)
_FSOUND_SetBufferSize(100 /* msec */ );
if (!_FSOUND_Init((int)srate, 2/*RTFM*/, 0))
goto LFailed2;
extern void streamcallback(FSOUND_STREAM*, void *buff, int len, int param);
fsound_stream = _FSOUND_Stream_Create(streamcallback, nchansOut*MaxSampsPerBuffer*2, FSOUND_LOOP_OFF | FSOUND_16BITS | \
(nchansOut==2 ? FSOUND_STEREO : FSOUND_MONO) \
| FSOUND_STREAMABLE | FSOUND_2D, (int)srate, 12345);
if (!fsound_stream)
goto LFailed2;
if (_FSOUND_Stream_Play(FSOUND_FREE, fsound_stream) == -1)
{
LFailed2:
printf("vss FMOD error: %s\n", FMOD_ErrorString(_FSOUND_GetError()));
FreeLibrary(fmod_dll);
goto LFailed;
}
goto LContinue;
}
LFailed:
if (vfMMIO)
cerr <<"vss failed to initialize MMIO. Can't output audio.\n";
else
cerr <<"vss failed to initialize FMOD/DirectSound. Can't output audio.\n";
liveaudio = 0;
#endif // VSS_WINDOWS
}
#ifndef VSS_MAC
LContinue:
#endif
{
const double Cdelt = 2./(double)wSoftclipLim;
const double Cdelt2 = Cdelt*Cdelt;
const double Cdelt3 = Cdelt*Cdelt2;
const double Aval = 1. - 2.*(1.-(wSoftclipLim/2.))/(wSoftclipLim/2.);
const double Bval = -2. + 3.*(1.-(wSoftclipLim/2.))/(wSoftclipLim/2.);
const double Cval = 1.;
double Cy = wSoftclipLim/2.;
double Cyi = Aval*Cdelt3 + Bval*Cdelt2 + Cval*Cdelt;
double Cyii = 6.*Aval*Cdelt3 + 2.*Bval*Cdelt2;
const double Cyiii = 6.*Aval*Cdelt3;
for (int m = 0; m < wSoftclipLim/2; ++m)
{
rgwSoftclip[m] = m;
rgwSoftclip[m + wSoftclipLim/2] = (int)Cy;
Cy += Cyi * wSoftclipLim / 2.;
Cyi += Cyii;
Cyii += Cyiii;
}
rgwSoftclip[wSoftclipLim] = 1;
}
vfWaitForReinit = false; // enable LiveTick()
#ifdef VSS_LINUX
#ifdef VSS_LINUX_UBUNTU
return liveaudio ? 0 : -1;
#else
if (!liveaudio)
{
if (fdDAC >= 0)
close(fdDAC);
fdDAC = -1;
}
return fdDAC;
#endif
#elif defined VSS_IRIX
return liveaudio ? alGetFD(alp) : -1;
#elif defined VSS_WINDOWS || defined VSS_MAC
return liveaudio ? 0 : -1;
#endif
}
void VSS_ResyncHardware()
{
#ifdef VSS_LINUX_UBUNTU
if (pcm_handle_read)
snd_pcm_drain(pcm_handle_read);
if (pcm_handle_write)
snd_pcm_drain(pcm_handle_write);
#endif
}
// How many samples can we compute without getting too far ahead?
int Scount()
{
#ifdef VSS_IRIX
return alGetFilled(alp);
#elif defined VSS_LINUX_UBUNTU
// # frames ready to capture or play (how far from xrun): snd_pcm_avail(), or cheap approximate snd_pcm_avail_update().
return pcm_handle_write ? snd_pcm_avail_update(pcm_handle_write) : 0;
#elif defined VSS_WINDOWS
return MaxSampsPerBuffer; // wild guess
#elif defined VSS_MAC
return MaxSampsPerBuffer; // wild guess
#endif
}
static float global_ampl = 1.0;
extern void VSS_SetGlobalAmplitude(float ampl) { global_ampl = ampl; }
extern float VSS_GetGlobalAmplitude() { return global_ampl; }
static bool fWantToResetsynth = false;
void Closesynth()
{
if (globs.liveaudio)
{
#ifdef VSS_IRIX
alClosePort(alp);
#elif defined VSS_LINUX_UBUNTU
if (pcm_handle_write) {
snd_pcm_drain(pcm_handle_write);
snd_pcm_close(pcm_handle_write);
}
if (pcm_handle_read) {
snd_pcm_drain(pcm_handle_read);
snd_pcm_close(pcm_handle_read);
}
#elif defined VSS_WINDOWS
if (vfCalledback)
{
//printf("Closesynth called during callback. dehr?! It's waiting.\n");
vfLiveTickPaused = 1; // Ack that FMOD callback needs to run.
//;; vfDie = true; // give callback a hint to go away already, eh.
while (vfCalledback) // wait for callback to go away
usleep(2000);
printf("Closesynth finished waiting.\n");;;;
}
if (vfMMIO)
{
areal_internal_TermAudio();
}
else
{
printf("in Closesynth, fWantToResetsynth==%d\n", fWantToResetsynth);;;;
if (fsound_stream)
{
if (vfCalledback)
printf("\n\n\nmaybe internal error!!! asdf\n");;;;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_Stream_Stop@4"); TESTIT
signed char (*_FSOUND_Stream_Stop)(FSOUND_STREAM*) = (signed char(*)(FSOUND_STREAM*))pfn;
_FSOUND_Stream_Stop(fsound_stream);
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_Stream_Close@4"); TESTIT
signed char (*_FSOUND_Stream_Close)(FSOUND_STREAM*) = (signed char(*)(FSOUND_STREAM*))pfn;
_FSOUND_Stream_Close(fsound_stream);
fsound_stream = NULL;
pfn = (int (*)())GetProcAddress(fmod_dll, "_FSOUND_Close@0"); TESTIT
void (*_FSOUND_Close)() = (void(*)())pfn;
_FSOUND_Close();
// don't bother... just messier for reset button.
// FreeLibrary(fmod_dll);
}
else
cerr << "vss internal error: Closesynth() called twice.\n";
}
#endif // VSS_WINDOWS
}
if (globs.fdOfile >= 0)
CloseOfile(globs.ofile);
usleep(100000);
}
void WantToResetsynth()
{
fWantToResetsynth = true;
}
static void MaybeResetsynth()
{
if (!fWantToResetsynth)
return;
Closesynth();
fWantToResetsynth = false;
(void)globs.Initsynth();
fWantToResetsynth = false;
}
int Synth(int n) {
const int nchans = Nchans();
#ifndef VSS_WINDOWS
MaybeResetsynth();
#endif
float k = global_ampl;
int i,j;
if (globs.liveaudio && fSoundIn)
{
#ifdef VSS_LINUX_UBUNTU
const int rc = snd_pcm_readi(pcm_handle_read, ibuf, n);
if (rc == -EPIPE) {
fprintf(stderr, "vss: input overrun.\n");
snd_pcm_prepare(pcm_handle_read);
} else if (rc < 0) {
fprintf(stderr, "vss: input error: %s\n", snd_strerror(rc));
} else if (rc != n) {
fprintf(stderr, "vss: input read only %d frames, not %d\n", rc, n);
}
#if 0
// Cheap VU meter for input.
auto p = 0; for (i=0; i<n*nchansIn; ++i) p = std::max(p, abs(ibuf[i]));
if (p != 0) printf("peak = %4d\n", p);
#endif
#elif defined VSS_IRIX
alReadFrames(alpin, ibuf, n);
#endif
#define REMOVE_DC
#ifndef REMOVE_DC
constexpr auto wDCOffset = 0;
#else
// Remove DC offset from input, assuming that input
// was roughly silent when VSS started.
static auto wDCOffset = 1<<20;
if (wDCOffset == 1<<20) {
wDCOffset = 0;
if (NchansIn() > 0) {
for (i=0; i<n*NchansIn(); ++i)
wDCOffset += ibuf[i];
wDCOffset /= -n*NchansIn();
} // Else, input failed to init.
}
#endif
for (i=0; i<n; ++i)
for (j=0; j<NchansIn(); ++j)
inpvecp[i*MaxNumChannels + j] =
(ibuf[i*NchansIn() + j] + wDCOffset) / 32768.0f;
/* 32768 not 32767, to stay >= -1. */
// So input to vss is [-1,1]. But output is [-32k,32k].
// That's ok, synthesis classes only see [-1,1].
}
// Stuff the output buffer.
ZeroFloats(outvecp, n * nchans);
for (auto alg: VAlgorithm::Generators)
alg->outputSamples(n, outvecp, nchans);
globs.SampleCount += n;
if (globs.fRemappedOutput)
{
static float temp[MaxSampsPerBuffer][MaxNumChannels];
for (i=0; i<n; i++)
for (j=0; j<globs.nchansOut; j++)
temp[i][j] = outvecp[i*nchans + globs.rgwRemappedOutput[j]];
for (i=0; i<n; i++)
for (j=0; j<globs.nchansOut; j++)
outvecp[i*globs.nchansOut + j] = temp[i][j];
}
ssp = sampbuff;
if (vfLimitClip)
{
/*
very fast gain reduction and a very slow gain increase.
if level gets within -2dB of clip, reduce gain "immediately"
(within 128 samples, = 3 msec at 44kHz) by 5dB.
else if >-5dB, reduce gain by .5dB per second
else if <-30 dB, increase gain by 0.1dB per second.
else if <-15dB, increase gain by .1dB per second
Boost is no more than 60dB, though (600 seconds).
Cut is no more than 200dB (120 msec).
slower pumping to keep it between -15dB and -5dB.
*/
static float zLimitdB = 0.;
static int fShouted = 0;
if (fShouted>0)
--fShouted;
float ampMax = 0.;
for (i=0; i < n*nchans; ++i)
{
float amp = outvecp[i];
if (amp > ampMax)
ampMax = amp;
}
float dBMax = dBFromScalar(ampMax * k * ScalarFromdB(zLimitdB) / 32768.);
// printf("ampMax = %.2f dBMax = %.2f\n", ampMax, dBMax);;
if (dBMax > -2.)
fprintf(stderr, "vss: avoiding hard clipping on output.\n");
float dBNew = 0.;
if (dBMax > 0.)
dBNew = -dBMax - 4.;
else if (dBMax > -2.)
dBNew = -5.;
else if (dBMax > -5.)
dBNew = -0.5 * globs.OneOverSR * MaxSampsPerBuffer;
// globs.OneOverSR * MaxSampsPerBuffer is seconds per buffer
else if (dBMax < -30.)
dBNew = 0.4 * globs.OneOverSR * MaxSampsPerBuffer;
else if (dBMax < -15.)
dBNew = 0.2 * globs.OneOverSR * MaxSampsPerBuffer;
zLimitdB += dBNew;
k *= ScalarFromdB(zLimitdB);
if (!fShouted)
{
if (zLimitdB < -3.)
fprintf(stderr, "vss: limiting output by %.1f dB\n",
-zLimitdB);
if (zLimitdB > 3.)
fprintf(stderr, "vss: boosting output by %.1f dB\n",
zLimitdB);
// report this not more than once every 10 seconds
fShouted = (int)(10.0 * globs.SampleRate/MaxSampsPerBuffer);
}
for (i=0; i < n*nchans; ++i)
{
float wAmpl = outvecp[i] * k;
*ssp++ = (short)wAmpl;
}
}
else if (vfSoftClip)
{
static int fShoutedSoft = 0;
if (fShoutedSoft>0)
--fShoutedSoft;
static int fShouted = 0;
if (fShouted>0)
--fShouted;
for (i=0; i<n*nchans; ++i)
{
int wT = (int)(outvecp[i] * k);
int wAmpl = abs(wT);
if (wAmpl > wSoftclipLim/2)
{
if (!fShoutedSoft && !fShouted)
{
fprintf(stderr, "vss: soft clipping.\n");
// report this not more than once every 10 seconds
fShoutedSoft = (int)(10.0 * globs.SampleRate/MaxSampsPerBuffer);
}
if (wAmpl > wSoftclipLim)
{
wAmpl = wSoftclipLim;
if (!fShouted)
{
fprintf(stderr, "vss: hard clipping (%.2f\n", fabs(outvecp[i]) * k / 32768);
// report this not more than once every 2 seconds
fShouted = (int)(2.0 * globs.SampleRate/MaxSampsPerBuffer);
fShoutedSoft = (int)(10.0 * globs.SampleRate/MaxSampsPerBuffer);
}
}
wT = (wT >= 0) ?
rgwSoftclip[wAmpl] :
-rgwSoftclip[wAmpl];
}
*ssp++ = wT;
}
}
else
{
static bool fFirstShout = true; // The first one might be a bogus NaN.
static int fShouted = 0;
if (fShouted>0)
--fShouted;
for (i=0; i < n*nchans; ++i) {
double wAmpl = outvecp[i] * k;
const auto pos = abs(wAmpl);
if (pos > 0.9995*32768) {
if (fShouted <= 0) {
if (fFirstShout) {
fFirstShout = false;
} else {
fprintf(stderr, "vss: hard clipping (%.2f)\n", pos/32768.);
// report this not more than once every 2 seconds
fShouted = 2.0 * globs.SampleRate/MaxSampsPerBuffer;
}
}
// Actually clip. Don't just wrap around (2021).
wAmpl = wAmpl > 0.0 ? 32767 : -32767;
}
*ssp++ = wAmpl;
}
}
{
int samps = ssp - sampbuff;
if (globs.liveaudio)
{
#ifdef VSS_IRIX
alWriteFrames(alp, sampbuff, samps/nchans);
alSetFillPoint(alp, (long)(qsize-latency));
#elif defined VSS_WINDOWS
// Tell streamcallback() where to find the samples.
vrgsCallback = sampbuff;
vcbCallback = samps*sizeof(short);
#elif defined VSS_LINUX_UBUNTU
// to sync: snd_pcm_delay() returns how much time will pass before a sample written now gets played.
{
signed short *ptr = sampbuff; // sampbuff is interleaved.
int cptr = samps / NchansOut();
while (cptr > 0) {
const auto err = snd_pcm_writei(pcm_handle_write, ptr, cptr);
if (err == -EAGAIN)
continue;
if (err < 0) {
if (xrun_recovery(pcm_handle_write, err) < 0) {
printf("vss failed to play: %s\n", snd_strerror(err));
return false;
}
break;
}
ptr += err * NchansOut(); // err is also # of frames written.
cptr -= err;
}
}
#else
#warning "Audio output unimplemented for this platform."
#endif // many platforms
}
if (globs.fdOfile >= 0 && globs.ofile_enabled)
{
// ALSA does this with the "tee device" tee:hw.
const int cb = samps*sizeof(short);
if (globs.vcbBufOfile)
{
if (globs.vibBufOfile + cb >= globs.vcbBufOfile)
{
// flush memory buffer
(void)!write(globs.fdOfile, globs.rgbBufOfile, globs.vibBufOfile);
globs.vibBufOfile=0;
}
// append to memory buffer
memcpy(globs.rgbBufOfile+globs.vibBufOfile, (const char*)sampbuff, cb);
globs.vibBufOfile += cb;
}
else
(void)!write(globs.fdOfile, sampbuff, cb);
}
}
return true;
}
#ifndef VSS_WINDOWS
int main(int argc,char *argv[])
{
return VSS_main(argc, argv);
}
#endif
static int initudp(int chan)
{
const auto sockfd = socket(AF_INET, SOCK_DGRAM, 0);
if (sockfd < 0)
return -1;
struct sockaddr_in serv_addr = {0};
serv_addr.sin_family = AF_INET;
serv_addr.sin_addr.s_addr = htonl(INADDR_ANY);
serv_addr.sin_port = htons(chan);
if (bind(sockfd, (struct sockaddr*)&serv_addr, sizeof serv_addr) < 0)
return -1;
fcntl(sockfd, F_SETFL, FNDELAY);
return sockfd;
}
static void closeudp(int sockfd)
{
close(sockfd);
}
const size_t MAXMESG = 500; // Small enough to avoid fragmentation when MTU is typically 1500 bytes.
char mbuf[MAXMESG];
int caught_sigint = 0;
#if defined VSS_IRIX_63_MIPS3 || defined VSS_LINUX || defined VSS_WINDOWS || defined VSS_MAC
#define SignalHandlerType int
#else
#define SignalHandlerType ...
#endif
void catch_sigint(SignalHandlerType)
{
caught_sigint = 1;
}
#ifdef VSS_IRIX
void doActors();
void doActorsCleanup();
void deleteActors();
#endif
static int viGear = 1;
enum { prndl_parked=0, prndl_low, prndl_drive }; // for viGear
bool FParked() { return viGear == prndl_parked; }
bool FDrive() { return viGear == prndl_drive; }
void VSS_SetGear(int iGear)
{
if (iGear != prndl_parked && iGear != prndl_low && iGear != prndl_drive)
return;
if (iGear != prndl_parked && viGear == prndl_parked)
{
// switching out of park: resynch (input and) output
VSS_ResyncHardware();
}
viGear = iGear;
}
#ifndef VSS_WINDOWS
struct pollfd vpfd; // vpfd.fd replaces the redundant udpDesc.sockfd
#endif
// Compute (c*MaxSampsPerBuffer) samples into the output buffer.
//
// wCatchUp seems like a very bad idea. It allows sample buffers
// to be computed far in advance of what can be written, and then
// the write() (or whatever call) has to block! Dehr?
static void doSynth(int r, int fForce=0, int wCatchUp=0)
{