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rtalsa.c
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rtalsa.c
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/*
rtalsa.c:
Copyright (C) 2005 Istvan Varga
(C) 2009 Andrés Cabrera, Clemens Ladisch
(C) 2012 Tito Latini
This file is part of Csound.
The Csound Library is free software; you can redistribute it
and/or modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
Csound 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 Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with Csound; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
02110-1301 USA
*/
#ifndef _ISOC99_SOURCE
#define _ISOC99_SOURCE 1
#endif
#ifndef _POSIX_C_SOURCE
#define _POSIX_C_SOURCE 200112L
#endif
#ifndef _DEFAULT_SOURCE
#define _DEFAULT_SOURCE 1
#endif
/* _BSD_SOURCE definition can be dropped once support for glibc < 2.19 is dropped */
#ifndef _BSD_SOURCE
#define _BSD_SOURCE 1
#endif
#include "csdl.h"
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/time.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/select.h>
#include <termios.h>
#include <errno.h>
#include <stdio.h>
#include <alsa/asoundlib.h>
#include <sched.h>
#include <unistd.h>
#include <signal.h>
#include <sys/mman.h>
#include <sys/resource.h>
#include "soundio.h"
/* Modified from BSD sources for strlcpy */
/*
* Copyright (c) 1998 Todd C. Miller <Todd.Miller@courtesan.com>
*
* Permission to use, copy, modify, and distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*/
/* modifed for speed -- JPff */
char *
strNcpy(char *dst, const char *src, size_t siz)
{
char *d = dst;
const char *s = src;
size_t n = siz;
/* Copy as many bytes as will fit or until NULL */
if (n != 0) {
while (--n != 0) {
if ((*d++ = *s++) == '\0')
break;
}
}
/* Not enough room in dst, add NUL */
if (n == 0) {
if (siz != 0)
*d = '\0'; /* NUL-terminate dst */
//while (*s++) ;
}
return dst; /* count does not include NUL */
}
#define MSG(csound, fmt, ...) { \
if(csound->GetMessageLevel(csound) || csound->GetDebug(csound)) { \
csound->Message(csound, fmt, __VA_ARGS__); \
} \
}
typedef struct devparams_ {
snd_pcm_t *handle; /* handle */
void *buf; /* sample conversion buffer */
char *device; /* device name */
int format; /* sample format */
int sampleSize; /* MYFLT sample frame size in bytes */
uint32_t srate; /* sample rate in Hz */
int nchns; /* number of channels */
int buffer_smps; /* buffer length in samples */
int period_smps; /* period time in samples */
/* playback sample conversion function */
void (*playconv)(int, MYFLT *, void *, int *);
/* record sample conversion function */
void (*rec_conv)(int, void *, MYFLT *);
int seed; /* random seed for dithering */
} DEVPARAMS;
#ifdef BUF_SIZE
#undef BUF_SIZE
#endif
#define BUF_SIZE 4096
typedef struct alsaMidiInputDevice_ {
unsigned char buf[BUF_SIZE];
snd_rawmidi_t *dev;
int bufpos, nbytes, datreq;
unsigned char prvStatus, dat1, dat2;
struct alsaMidiInputDevice_ *next;
} alsaMidiInputDevice;
typedef struct midiDevFile_ {
unsigned char buf[BUF_SIZE];
int fd;
int bufpos, nbytes, datreq;
unsigned char prvStatus, dat1, dat2;
} midiDevFile;
typedef struct alsaseqMidi_ {
snd_seq_t *seq;
snd_midi_event_t *mev;
snd_seq_event_t sev;
snd_seq_client_info_t *cinfo;
snd_seq_port_info_t *pinfo;
} alsaseqMidi;
static const unsigned char dataBytes[16] = {
0, 0, 0, 0, 0, 0, 0, 0, 2, 2, 2, 2, 1, 1, 2, 0
};
int set_scheduler_priority(CSOUND *csound, int priority)
{
struct sched_param p;
memset(&p, 0, sizeof(struct sched_param));
if (UNLIKELY(priority < -20 || priority > sched_get_priority_max(SCHED_RR))) {
csound->Message(csound,
Str("--scheduler: invalid priority value; "
"the allowed range is:"));
csound->Message(csound,Str(" -20 to -1: set nice level"));
csound->Message(csound,Str(" 0: normal scheduling, "
"but lock memory"));
csound->Message(csound,Str(" 1 to %d: SCHED_RR with the specified "
"priority (DANGEROUS)"),
sched_get_priority_max(SCHED_RR));
return -1;
}
/* set scheduling policy and priority */
if (priority > 0) {
p.sched_priority = priority;
if (UNLIKELY(sched_setscheduler(0, SCHED_RR, &p) != 0)) {
csound->Message(csound,
Str("csound: cannot set scheduling policy to SCHED_RR"));
}
else csound->Message(csound,
Str("csound: setting scheduling policy to SCHED_RR\n"));
}
else {
/* nice requested */
if (UNLIKELY(setpriority(PRIO_PROCESS, 0, priority) != 0)) {
csound->Message(csound, Str("csound: cannot set nice level to %d"),
priority);
}
}
return 0;
}
/* sample conversion routines for playback */
static void MYFLT_to_short(int nSmps, MYFLT *inBuf, int16_t *outBuf, int *seed)
{
MYFLT tmp_f;
int tmp_i;
int n;
for (n=0; n<nSmps; n++) {
int rnd = (((*seed) * 15625) + 1) & 0xFFFF;
*seed = (((rnd) * 15625) + 1) & 0xFFFF;
rnd += *seed; /* triangular distribution */
tmp_f = (MYFLT) ((rnd>>1) - 0x8000) * (FL(1.0) / (MYFLT) 0x10000);
tmp_f += inBuf[n] * (MYFLT) 0x8000;
#ifndef USE_DOUBLE
tmp_i = (int) lrintf(tmp_f);
#else
tmp_i = (int) lrint(tmp_f);
#endif
if (tmp_i < -0x8000) tmp_i = -0x8000;
if (tmp_i > 0x7FFF) tmp_i = 0x7FFF;
outBuf[n] = (int16_t) tmp_i;
}
}
static void MYFLT_to_short_u(int nSmps, MYFLT *inBuf, int16_t *outBuf, int *seed)
{
MYFLT tmp_f;
int tmp_i;
int n;
for (n=0; n<nSmps; n++) {
int rnd = (((*seed) * 15625) + 1) & 0xFFFF;
*seed = rnd;
tmp_f = (MYFLT) (rnd - 0x8000) * (FL(1.0) / (MYFLT) 0x10000);
tmp_f += inBuf[n] * (MYFLT) 0x8000;
#ifndef USE_DOUBLE
tmp_i = (int) lrintf(tmp_f);
#else
tmp_i = (int) lrint(tmp_f);
#endif
if (tmp_i < -0x8000) tmp_i = -0x8000;
if (tmp_i > 0x7FFF) tmp_i = 0x7FFF;
outBuf[n] = (int16_t) tmp_i;
}
}
static void MYFLT_to_short_no_dither(int nSmps, MYFLT *inBuf,
int16_t *outBuf, int *seed)
{
IGN(seed);
MYFLT tmp_f;
int tmp_i;
int n;
for (n=0; n<nSmps; n++) {
tmp_f = inBuf[n] * (MYFLT) 0x8000;
#ifndef USE_DOUBLE
tmp_i = (int) lrintf(tmp_f);
#else
tmp_i = (int) lrint(tmp_f);
#endif
if (tmp_i < -0x8000) tmp_i = -0x8000;
if (tmp_i > 0x7FFF) tmp_i = 0x7FFF;
outBuf[n] = (int16_t) tmp_i;
}
}
static void MYFLT_to_long(int nSmps, MYFLT *inBuf, int32_t *outBuf, int *seed)
{
MYFLT tmp_f;
int64_t tmp_i;
(void) seed;
int n;
for (n=0; n<nSmps; n++) {
tmp_f = inBuf[n] * (MYFLT) 0x80000000UL;
#ifndef USE_DOUBLE
tmp_i = (int64_t) llrintf(tmp_f);
#else
tmp_i = (int64_t) llrint(tmp_f);
#endif
if (tmp_i < -((int64_t) 0x80000000UL))
tmp_i = -((int64_t) 0x80000000UL);
if (tmp_i > (int64_t) 0x7FFFFFFF) tmp_i = (int64_t) 0x7FFFFFFF;
outBuf[n] = (int32_t) tmp_i;
}
}
static void MYFLT_to_float(int nSmps, MYFLT *inBuf, float *outBuf, int *seed)
{
(void) seed;
int n;
for (n=0; n<nSmps; n++)
outBuf[n] = (float) inBuf[n];
}
/* sample conversion routines for recording */
static void short_to_MYFLT(int nSmps, int16_t *inBuf, MYFLT *outBuf)
{
int n;
MYFLT adjust = FL(1.0) / (MYFLT) 0x8000;
for (n=0; n<nSmps; n++)
outBuf[n] = (MYFLT) inBuf[n] * adjust;
}
static void long_to_MYFLT(int nSmps, int32_t *inBuf, MYFLT *outBuf)
{
int n;
MYFLT adjust = FL(1.0) / (MYFLT) 0x80000000UL;
for (n=0; n<nSmps; n++)
outBuf[n] = (MYFLT) inBuf[n] * adjust;
}
static void float_to_MYFLT(int nSmps, float *inBuf, MYFLT *outBuf)
{
int n;
for (n=0; n<nSmps; n++)
outBuf[n] = (MYFLT) inBuf[n];
}
/* select sample format */
static snd_pcm_format_t set_format(void (**convFunc)(void), int csound_format,
int play, int csound_dither)
{
int16 endian_test = 0x1234;
(*convFunc) = NULL;
/* select conversion routine */
switch (csound_format) {
case AE_SHORT:
if (play) {
if (csound_dither==1)
*convFunc = (void (*)(void)) MYFLT_to_short;
else if (csound_dither==2)
*convFunc = (void (*)(void)) MYFLT_to_short_u;
else
*convFunc = (void (*)(void)) MYFLT_to_short_no_dither;
}
else
*convFunc = (void (*)(void)) short_to_MYFLT;
break;
case AE_LONG:
if (play)
*convFunc = (void (*)(void)) MYFLT_to_long;
else
*convFunc = (void (*)(void)) long_to_MYFLT;
break;
case AE_FLOAT:
if (play)
*convFunc = (void (*)(void)) MYFLT_to_float;
else
*convFunc = (void (*)(void)) float_to_MYFLT;
break;
}
if (*((unsigned char*) (&endian_test)) == (unsigned char) 0x34) {
/* little-endian */
switch (csound_format) {
case AE_SHORT: return SND_PCM_FORMAT_S16_LE;
case AE_LONG: return SND_PCM_FORMAT_S32_LE;
case AE_FLOAT: return SND_PCM_FORMAT_FLOAT_LE;
}
}
else {
/* big-endian */
switch (csound_format) {
case AE_SHORT: return SND_PCM_FORMAT_S16_BE;
case AE_LONG: return SND_PCM_FORMAT_S32_BE;
case AE_FLOAT: return SND_PCM_FORMAT_FLOAT_BE;
}
}
return SND_PCM_FORMAT_UNKNOWN;
}
static void DAC_channels(CSOUND *csound, int chans){
int *dachans = (int *) csound->QueryGlobalVariable(csound, "_DAC_CHANNELS_");
if (dachans == NULL) {
if (csound->CreateGlobalVariable(csound, "_DAC_CHANNELS_",
sizeof(int)) != 0)
return;
dachans = (int *) csound->QueryGlobalVariable(csound, "_DAC_CHANNELS_");
*dachans = chans;
}
}
static void ADC_channels(CSOUND *csound, int chans){
int *dachans = (int *) csound->QueryGlobalVariable(csound, "_ADC_CHANNELS_");
if (dachans == NULL) {
if (csound->CreateGlobalVariable(csound, "_ADC_CHANNELS_",
sizeof(int)) != 0)
return;
dachans = (int *) csound->QueryGlobalVariable(csound, "_ADC_CHANNELS_");
*dachans = chans;
}
}
/* set up audio device */
#define MSGLEN (512)
static int set_device_params(CSOUND *csound, DEVPARAMS *dev, int play)
{
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
snd_pcm_format_t alsaFmt;
int err, n, alloc_smps;
CSOUND *p = csound;
char *devName, msg[MSGLEN];
dev->buf = NULL;
snd_pcm_hw_params_alloca(&hw_params);
snd_pcm_sw_params_alloca(&sw_params);
/* open the device */
if (dev->device == NULL || dev->device[0] == '\0')
devName = "default";
else
devName = dev->device;
err = snd_pcm_open(&(dev->handle), devName,
(play ? SND_PCM_STREAM_PLAYBACK
: SND_PCM_STREAM_CAPTURE), 0);
if (UNLIKELY(err < 0)) {
if (play)
p->ErrorMsg(p, Str(" *** Cannot open device '%s' for audio output: %s"),
devName, snd_strerror(err));
else
p->ErrorMsg(p, Str(" *** Cannot open device '%s' for audio input: %s"),
devName, snd_strerror(err));
return -1;
}
/* allocate hardware and software parameters */
if (UNLIKELY(snd_pcm_hw_params_any(dev->handle, hw_params) < 0)) {
strNcpy(msg, Str("No real-time audio configurations found"), MSGLEN);
goto err_return_msg;
}
/*=======================*/
unsigned int hwchns;
if (UNLIKELY(snd_pcm_hw_params_get_channels_max(hw_params, &hwchns) < 0)) {
strNcpy(msg, Str("Could not retrieve max number of channels"), MSGLEN);
goto err_return_msg;
}
if(play) {
DAC_channels(csound,hwchns);
}
else ADC_channels(csound,hwchns);
/*=========================*/
/* now set the various hardware parameters: */
/* access method, */
if (UNLIKELY(snd_pcm_hw_params_set_access(dev->handle, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED) < 0)) {
strNcpy(msg, Str("Error setting access type for soundcard"), MSGLEN);
goto err_return_msg;
}
/* sample format, */
alsaFmt = SND_PCM_FORMAT_UNKNOWN;
if(dev->srate == 0) dev->format = AE_FLOAT;
dev->sampleSize = (int) sizeof(MYFLT) * dev->nchns;
{
void (*fp)(void) = NULL;
alsaFmt = set_format(&fp, dev->format, play, csound->GetDitherMode(csound));
if (play) dev->playconv = (void (*)(int, MYFLT*, void*, int*)) fp;
else dev->rec_conv = (void (*)(int, void*, MYFLT*)) fp;
}
if (UNLIKELY(alsaFmt == SND_PCM_FORMAT_UNKNOWN)) {
strNcpy(msg, Str("Unknown sample format.\n *** Only 16-bit and 32-bit "
"integers, and 32-bit floats are supported."), MSGLEN);
goto err_return_msg;
}
if (UNLIKELY(snd_pcm_hw_params_set_format(dev->handle, hw_params, alsaFmt)<0)) {
strNcpy(msg,
Str("Unable to set requested sample format on soundcard"),MSGLEN);
goto err_return_msg;
}
/* number of channels, */
if (UNLIKELY(snd_pcm_hw_params_set_channels(dev->handle, hw_params,
(unsigned int) dev->nchns) < 0)) {
strNcpy(msg, Str("Unable to set number of channels on soundcard"), MSGLEN);
goto err_return_msg;
}
/* sample rate, (patched for sound cards that object to fixed rate) */
{
unsigned int target;
if(dev->srate == 0) {
// VL 2-4-2019 this code gets HW sr for use in Csound.
unsigned int hwsr;
snd_pcm_hw_params_t *pms;
snd_pcm_hw_params_alloca(&pms);
snd_pcm_hw_params_any(dev->handle, pms);
snd_pcm_hw_params_get_rate(pms, &hwsr, 0);
if(hwsr == 0) hwsr = 44100;
csound->system_sr(csound, hwsr);
target = dev->srate = hwsr;
MSG(p, "alsa hw sampling rate: %d\n", hwsr);
}
else target = (unsigned int) dev->srate;
if (UNLIKELY(snd_pcm_hw_params_set_rate_near(dev->handle,
hw_params,
(unsigned int *) &dev->srate, 0)
< 0)) {
strNcpy(msg, Str("Unable to set sample rate on soundcard"), MSGLEN);
goto err_return_msg;
}
if (dev->srate!=target)
p->MessageS(p, CSOUNDMSG_WARNING, Str(" *** rate set to %d\n"), dev->srate);
csound->system_sr(csound, dev->srate);
}
/* buffer size, */
if (dev->buffer_smps == 0)
dev->buffer_smps = 1024;
else if (dev->buffer_smps < 16)
dev->buffer_smps = 16;
{
snd_pcm_uframes_t nn = (snd_pcm_uframes_t) dev->buffer_smps;
err = snd_pcm_hw_params_set_buffer_size_near(dev->handle, hw_params, &nn);
if (err < 0 || (int) nn != dev->buffer_smps) {
if (UNLIKELY(err >= 0)) {
p->Message(p, Str("ALSA: -B %d not allowed on this device; "
"using %d instead\n"), dev->buffer_smps, (int) nn);
dev->buffer_smps=nn;
}
}
}
/* and period size */
alloc_smps = dev->period_smps;
if (dev->period_smps == 0)
dev->period_smps = 256;
else if (dev->period_smps < 8)
dev->period_smps = 8;
else if (dev->period_smps > (dev->buffer_smps >> 1))
dev->period_smps = (dev->buffer_smps >> 1);
if (alloc_smps < dev->period_smps) /* make sure that enough memory */
alloc_smps = dev->period_smps; /* is allocated for the buffer */
{
snd_pcm_uframes_t nn = (snd_pcm_uframes_t) dev->period_smps;
int dir = 0;
err = snd_pcm_hw_params_set_period_size_near(dev->handle, hw_params, &nn,
&dir);
if (err < 0 || (int) nn != dev->period_smps) {
if (UNLIKELY(err >= 0)) {
p->Message(p, Str("ALSA: -b %d not allowed on this device; "
"using %d instead\n"), dev->period_smps, (int) nn);
dev->period_smps=nn;
}
}
}
/* set up device according to the above parameters */
if (UNLIKELY(snd_pcm_hw_params(dev->handle, hw_params) < 0)) {
strNcpy(msg,
Str("Error setting hardware parameters for real-time audio"),
MSGLEN);
goto err_return_msg;
}
/* print settings */
if (p->GetMessageLevel(p) != 0)
p->Message(p, Str("ALSA %s: total buffer size: %d, period size: %d\n"),
(play ? "output" : "input"),
dev->buffer_smps, dev->period_smps /*, dev->srate*/);
/* now set software parameters */
n = (play ? dev->buffer_smps : 1);
if (UNLIKELY(snd_pcm_sw_params_current(dev->handle, sw_params) < 0 ||
snd_pcm_sw_params_set_start_threshold(dev->handle, sw_params,
(snd_pcm_uframes_t) n) < 0 ||
snd_pcm_sw_params_set_avail_min(dev->handle, sw_params,
dev->period_smps) < 0 ||
/* snd_pcm_sw_params_set_xfer_align(dev->handle, sw_params, 1) < 0 || */
snd_pcm_sw_params(dev->handle, sw_params) < 0)) {
strNcpy(msg,
Str("Error setting software parameters for real-time audio"),MSGLEN);
goto err_return_msg;
}
/* allocate memory for sample conversion buffer */
n = (dev->format == AE_SHORT ? 2 : 4) * dev->nchns * alloc_smps;
dev->buf = (void*) csound->Malloc(csound, (size_t) n);
if (UNLIKELY(dev->buf == NULL)) {
strNcpy(msg, Str("Memory allocation failure"),MSGLEN);
goto err_return_msg;
}
memset(dev->buf, 0, (size_t) n);
/* device successfully opened */
return 0;
err_return_msg:
p->MessageS(p, CSOUNDMSG_ERROR, " *** %s\n", msg);
snd_pcm_close(dev->handle);
return -1;
}
static void list_devices(CSOUND *csound)
{
FILE * f = fopen("/proc/asound/pcm", "r");
/*file presents this format:
02-00: Analog PCM : Mona : playback 6 : capture 4*/
char *line, *line_;
line = (char *) csound->Calloc (csound, 128* sizeof(char));
line_ = (char *) csound->Calloc (csound, 128* sizeof(char));
char card_[] = " ";
char num_[] = " ";
char *temp;
if (f) {
char *th;
while (fgets(line, 128, f)) { /* Read one line*/
strcpy(line_, line);
temp = strtok_r (line, "-", &th);
strncpy (card_, temp, 2);
temp = strtok_r (NULL, ":", &th);
strncpy (num_, temp, 2);
int card = atoi (card_);
int num = atoi (num_);
temp = strchr (line_, ':');
if (temp)
temp = temp + 2;
/* name contains spaces at the beginning and the end.
And line return at the end*/
csound->Message(csound, " \"hw:%i,%i\" - %s",card, num, temp );
}
fclose(f);
}
csound->Free(csound, line);
csound->Free(csound, line_);
}
static void trim_trailing_whitespace(char *s) {
int i = 0, index = -1;
while(s[i] != '\0') {
if(s[i] != ' ' && s[i] != '\t' && s[i] != '\n')
index = i;
i++;
}
s[index+1] = '\0';
}
int listDevices(CSOUND *csound, CS_AUDIODEVICE *list, int isOutput){
IGN(csound);
FILE * f = fopen("/proc/asound/pcm", "r");
/*file presents this format:
02-00: Analog PCM : Mona : playback 6 : capture 4*/
char line[128], line_[128];
char card_[] = " ";
char num_[] = " ";
char *temp;
char tmp[64];
int n =0;
memset(line, '\0', 128); memset(line_, '\0', 128);
if (f) {
char *th;
while (fgets(line, 128, f)) { /* Read one line*/
strcpy(line_, line);
temp = strtok_r (line, "-", &th);
if (temp==NULL) {
fclose(f);
return 0;
}
strncpy (card_, temp, 2);
temp = strtok_r (NULL, ":", &th);
if (temp==NULL) {
fclose(f);
return 0;
}
strncpy (num_, temp, 2);
int card = atoi (card_);
int num = atoi (num_);
temp = strchr (line_, ':');
if (temp) {
temp = temp + 2;
} else {
fclose(f);
return 0;
}
if (list != NULL) {
/* for some reason, there appears to be a memory
problem if we try to copy more than 10 chars,
even though list[n].device_name is 64 chars long */
strNcpy(list[n].device_name, temp, 63);
trim_trailing_whitespace(list[n].device_name);
//list[n].device_name[10] = '\0';
snprintf(tmp, 64, "%shw:%i,%i", isOutput ? "dac:" : "adc:", card, num);
strNcpy(list[n].device_id, tmp, 16);
list[n].max_nchnls = -1;
list[n].isOutput = isOutput;
}
n++;
}
fclose(f);
}
return n;
}
static int open_device(CSOUND *csound, const csRtAudioParams *parm, int play)
{ DEVPARAMS *dev;
void **userDataPtr;
int retval;
userDataPtr = (play ? (void**) csound->GetRtPlayUserData(csound)
: (void**) csound->GetRtRecordUserData(csound));
/* check if the device is already opened */
if (*userDataPtr != NULL)
return 0;
if (UNLIKELY(parm->devNum != 1024)) {
csound->ErrorMsg(csound, Str(" *** ALSA: must specify a device name, "
"not a number (e.g. -odac:hw:0,0)"));
list_devices(csound);
return -1;
}
/* allocate structure */
dev = (DEVPARAMS*) csound->Malloc(csound, sizeof(DEVPARAMS));
if (UNLIKELY(dev == NULL)) {
csound->ErrorMsg(csound, Str(" *** ALSA: %s: memory allocation failure"),
(play ? "playopen" : "recopen"));
return -1;
}
*userDataPtr = (void*) dev;
memset(dev, 0, sizeof(DEVPARAMS));
/* set up parameters */
dev->handle = (snd_pcm_t*) NULL;
dev->buf = NULL;
dev->device = parm->devName;
dev->format = parm->sampleFormat;
dev->sampleSize = 1;
dev->srate = (int) (parm->sampleRate > 0 ? parm->sampleRate + 0.5f : 0);
dev->nchns = parm->nChannels;
dev->period_smps = parm->bufSamp_SW;
dev->playconv = (void (*)(int, MYFLT*, void*, int*)) NULL;
dev->rec_conv = (void (*)(int, void*, MYFLT*)) NULL;
dev->seed = 1;
/* open device */
retval = set_device_params(csound, dev, play);
if (retval != 0) {
csound->Free(csound,dev);
*userDataPtr = NULL;
}
return retval;
}
/* open for audio input */
static int recopen_(CSOUND *csound, const csRtAudioParams *parm)
{
return open_device(csound, parm, 0);
}
/* open for audio output */
static int playopen_(CSOUND *csound, const csRtAudioParams *parm)
{
return open_device(csound, parm, 1);
}
/* get samples from ADC */
#ifdef warning
#undef warning
#endif
#define warning(x) { \
if (UNLIKELY(csound->GetMessageLevel(csound) & 4)) \
csound->Warning(csound, Str(x)); \
}
static int rtrecord_(CSOUND *csound, MYFLT *inbuf, int nbytes)
{
DEVPARAMS *dev;
int n, m, err;
dev = (DEVPARAMS*) *(csound->GetRtRecordUserData(csound));
if (dev->handle == NULL) {
/* no device, return zero samples */
memset(inbuf, 0, (size_t) nbytes);
return nbytes;
}
/* calculate the number of samples to record */
n = nbytes / dev->sampleSize;
m = 0;
while (n) {
err = (int) snd_pcm_readi(dev->handle, dev->buf, (snd_pcm_uframes_t) n);
if (err >= 0) {
n -= err; m += err; continue;
}
/* handle I/O errors */
if (UNLIKELY(err == -EPIPE)) {
/* buffer underrun */
warning(Str("Buffer overrun in real-time audio input")); /* complain */
if (snd_pcm_prepare(dev->handle) >= 0) continue;
}
else if (err == -ESTRPIPE) {
/* suspend */
warning(Str("Real-time audio input suspended"));
while (snd_pcm_resume(dev->handle) == -EAGAIN) sleep(1);
if (snd_pcm_prepare(dev->handle) >= 0) continue;
}
/* could not recover from error */
csound->ErrorMsg(csound,
Str("Error reading data from audio input device"));
snd_pcm_close(dev->handle);
dev->handle = NULL;
break;
}
/* convert samples to MYFLT */
dev->rec_conv(m * dev->nchns, dev->buf, inbuf);
return (m * dev->sampleSize);
}
/* put samples to DAC */
static void rtplay_(CSOUND *csound, const MYFLT *outbuf, int nbytes)
{
DEVPARAMS *dev;
int n, err;
dev = (DEVPARAMS*) *(csound->GetRtPlayUserData(csound));
if (dev->handle == NULL)
return;
/* calculate the number of samples to play */
n = nbytes / dev->sampleSize;
/* convert samples from MYFLT */
dev->playconv(n * dev->nchns, (MYFLT*) outbuf, dev->buf, &(dev->seed));
while (n) {
err = (int) snd_pcm_writei(dev->handle, dev->buf, (snd_pcm_uframes_t) n);
if (err >= 0) {
n -= err; continue;
}
/* handle I/O errors */
if (err == -EPIPE) {
/* buffer underrun */
warning(Str("Buffer underrun in real-time audio output")); /* complain */
if (snd_pcm_prepare(dev->handle) >= 0) continue;
}
else if (err == -ESTRPIPE) {
/* suspend */
warning(Str("Real-time audio output suspended"));
while (snd_pcm_resume(dev->handle) == -EAGAIN) sleep(1);
if (snd_pcm_prepare(dev->handle) >= 0) continue;
}
/* could not recover from error */
csound->ErrorMsg(csound,
Str("Error writing data to audio output device"));
snd_pcm_close(dev->handle);
dev->handle = NULL;
break;
}
}
/* close the I/O device entirely */
/* called only when both complete */
static void rtclose_(CSOUND *csound)
{
DEVPARAMS *dev;
dev = (DEVPARAMS*) (*(csound->GetRtRecordUserData(csound)));
if (dev != NULL) {
*(csound->GetRtRecordUserData(csound)) = NULL;
if (dev->handle != NULL)
snd_pcm_close(dev->handle);
if (dev->buf != NULL)
csound->Free(csound, dev->buf);
csound->Free(csound,dev);
}
dev = (DEVPARAMS*) (*(csound->GetRtPlayUserData(csound)));
if (dev != NULL) {
*(csound->GetRtPlayUserData(csound)) = NULL;
if (dev->handle != NULL)
snd_pcm_close(dev->handle);
if (dev->buf != NULL)
csound->Free(csound, dev->buf);
csound->Free(csound,dev);
}
}
static alsaMidiInputDevice* open_midi_device(CSOUND *csound, const char *s)
{
int err;
alsaMidiInputDevice *dev;
dev = (alsaMidiInputDevice*) csound->Malloc(csound,
sizeof(alsaMidiInputDevice));
if (UNLIKELY(dev == NULL)) {
csound->ErrorMsg(csound, Str("ALSA MIDI: memory allocation failure"));
return dev;
}
memset(dev, 0, sizeof(alsaMidiInputDevice));
err = snd_rawmidi_open(&(dev->dev), NULL, s, SND_RAWMIDI_NONBLOCK);
if (UNLIKELY(err != 0)) {
csound->ErrorMsg(csound,
Str("ALSA: error opening MIDI input device: '%s'"), s);
csound->Free(csound,dev);
return NULL;
}
MSG(csound, Str("ALSA: opened MIDI input device '%s'\n"), s);
return dev;
}
// This is the function which contains code from amidi
static int midi_in_open(CSOUND *csound, void **userData, const char *devName)
{
alsaMidiInputDevice *dev = NULL, *newdev, *olddev;
//const char *s = "hw:0,0";
int card;
int device;
snd_ctl_t *ctl;
char* name;
int numdevs = 0;
name = (char *) csound->Calloc(csound, 32* sizeof(char));
(*userData) = NULL;
olddev = NULL;
if (UNLIKELY(devName==NULL)) {
csound->Message(csound, Str("ALSA midi: no string\n"));
exit(1); /* what should happen here???????? */
}
else if (devName[0] == 'a') {
if(csound->GetMessageLevel(csound) || csound->GetDebug(csound))
csound->Message(csound, Str("ALSA midi: Using all devices.\n"));
card = -1;
if (snd_card_next(&card) >= 0 && card >= 0) {
do {
snprintf(name, 32, "hw:%d", card);
if (snd_ctl_open(&ctl, name, 0) >= 0) {
device = -1;
for (;;) {
if (snd_ctl_rawmidi_next_device(ctl, &device) < 0) {
break;
}
if (device < 0) {
break;
}
snprintf(name, 32, "hw:%d,%d", card, device);
newdev = open_midi_device(csound, name);
if (LIKELY(newdev != NULL)) { /* Device opened successfully */
numdevs++;
if (olddev != NULL) {
olddev->next = newdev;
}
else { /* First Device */
dev = newdev;
}
olddev = newdev;
newdev = NULL;
}
else { /* Device couldn't be opened */
csound->Message(csound,
Str("ALSA midi: Error opening device: %s\n"),
name);
}
}
}
if (snd_card_next(&card) < 0)
break;
} while (card >= 0);
snd_ctl_close(ctl);
}
}
else if (devName[0] != '\0') {
dev = open_midi_device(csound, devName);
if (dev == NULL) {
csound->Free(csound, name);
return -1;
}
numdevs = 1;
}
csound->Free(csound, name);
if (UNLIKELY(numdevs == 0)) {
csound->ErrorMsg(csound, Str("ALSA midi: No devices found.\n"));
*userData = NULL;
}
else {
*userData = (void*) dev;
}
return 0;
}
static int midi_in_read(CSOUND *csound,
void *userData, unsigned char *buf, int nbytes)
{
alsaMidiInputDevice *dev = (alsaMidiInputDevice*) userData;
int bufpos = 0;
unsigned char c;
IGN(csound);
if (!dev) { /* No devices */
/* fprintf(stderr, "No devices!"); */
return 0;
}
/* (void) csound; */
dev->bufpos = 0;
while (dev && dev->dev) {
while ((nbytes - bufpos) >= 3) {
if (dev->bufpos >= dev->nbytes) { /* read from device */
int n = (int) snd_rawmidi_read(dev->dev, &(dev->buf[0]), BUF_SIZE);
dev->bufpos = 0;
if (n <= 0) { /* until there is no more data left */
dev->nbytes = 0;
break;
}
dev->nbytes = n;
}
c = dev->buf[dev->bufpos++];
if (c >= (unsigned char) 0xF8) { /* real time message */
buf[bufpos++] = c;
continue;
}
if (c == (unsigned char) 0xF7) /* end of system exclusive */
c = dev->prvStatus;