diff --git a/extra/sigmund~/sigmund~.c b/extra/sigmund~/sigmund~.c
index 03e4ab550d9b7a38b95a362acac23acb3812a8f3..776696019db1b356ce625f24726b6752e77f8518 100644
--- a/extra/sigmund~/sigmund~.c
+++ b/extra/sigmund~/sigmund~.c
@@ -1457,7 +1457,7 @@ static void *sigmund_new(t_symbol *s, long ac, t_atom *av)
dsp_setup((t_pxobject *)x, 1);
object_obex_store(x, gensym("dumpout"), outlet_new(x, NULL));
- for (i = 0; i < ac; i++) FIXME
+ for (i = 0; i < ac; i++)
if (av[i].a_type == A_SYM)
{
char *s = av[i].a_w.w_sym->s_name;
diff --git a/src/s_audio_alsa.c b/src/s_audio_alsa.c
index 6733182da80a338c907432396d7328d618126a76..dad2c86f5213592b3e92874cd612131243361d06 100644
--- a/src/s_audio_alsa.c
+++ b/src/s_audio_alsa.c
@@ -25,6 +25,7 @@
#include <sched.h>
#include <sys/mman.h>
#include "s_audio_alsa.h"
+#include <endian.h>
/* Defines */
#define DEBUG(x) x
@@ -117,17 +118,33 @@ static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate,
if (err < 0)
return (-1);
check_error(err, "snd_pcm_hw_params_set_access");
+#if 0 /* enable this to print out which formats are available */
+ {
+ int i;
+ for (i = 0; i <= SND_PCM_FORMAT_LAST; i++)
+ fprintf(stderr, "%d -> %d\n",
+ i, snd_pcm_hw_params_test_format(dev->a_handle, hw_params, i));
+ }
+#endif
/* Try to set 32 bit format first */
err = snd_pcm_hw_params_set_format(dev->a_handle,
hw_params, SND_PCM_FORMAT_S32);
if (err < 0)
{
/* fprintf(stderr,
- "PD-ALSA: 32 bit format not available - using 16\n"); */
+ "PD-ALSA: 32 bit format not available - trying 24\n"); */
err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,
- SND_PCM_FORMAT_S16);
- check_error(err, "snd_pcm_hw_params_set_format");
- dev->a_sampwidth = 2;
+ SND_PCM_FORMAT_S24_3LE);
+ if (err < 0)
+ {
+ /* fprintf(stderr,
+ "PD-ALSA: 32/24 bit format not available - using 16\n"); */
+ err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,
+ SND_PCM_FORMAT_S16);
+ check_error(err, "snd_pcm_hw_params_set_format");
+ dev->a_sampwidth = 2;
+ }
+ else dev->a_sampwidth = 3;
}
else dev->a_sampwidth = 4;
@@ -436,7 +453,32 @@ int alsa_send_dacs(void)
for (j = ch, k = DEFDACBLKSIZE; k--; j += thisdevchans)
((t_alsa_sample32 *)alsa_snd_buf)[j] = 0;
}
- else
+ else if (alsa_outdev[iodev].a_sampwidth == 3)
+ {
+ for (i = 0; i < chans; i++, ch++, fp1 += DEFDACBLKSIZE)
+ for (j = ch, k = DEFDACBLKSIZE, fp2 = fp1; k--;
+ j += thisdevchans, fp2++)
+ {
+ int s = *fp2 * 8388352.;
+ if (s > 8388351)
+ s = 8388351;
+ else if (s < -8388351)
+ s = -8388351;
+#if BYTE_ORDER == LITTLE_ENDIAN
+ ((char *)(alsa_snd_buf))[3*j] = (s & 255);
+ ((char *)(alsa_snd_buf))[3*j+1] = ((s>>8) & 255);
+ ((char *)(alsa_snd_buf))[3*j+2] = ((s>>16) & 255);
+#else
+ fprintf(stderr("big endian 24-bit not supported");
+#endif
+ }
+ for (; i < thisdevchans; i++, ch++)
+ for (j = ch, k = DEFDACBLKSIZE; k--; j += thisdevchans)
+ ((char *)(alsa_snd_buf))[3*j] =
+ ((char *)(alsa_snd_buf))[3*j+1] =
+ ((char *)(alsa_snd_buf))[3*j+2] = 0;
+ }
+ else /* 16 bit samples */
{
for (i = 0; i < chans; i++, ch++, fp1 += DEFDACBLKSIZE)
for (j = ch, k = DEFDACBLKSIZE, fp2 = fp1; k--;
@@ -525,6 +567,23 @@ int alsa_send_dacs(void)
* (1./ INT32_MAX);
}
}
+ else if (alsa_indev[iodev].a_sampwidth == 3)
+ {
+#if BYTE_ORDER == LITTLE_ENDIAN
+ for (i = 0; i < chans; i++, ch++, fp1 += DEFDACBLKSIZE)
+ {
+ for (j = ch, k = DEFDACBLKSIZE, fp2 = fp1; k--;
+ j += thisdevchans, fp2++)
+ *fp2 = ((float) (
+ (((unsigned char *)alsa_snd_buf)[3*j] << 8)
+ | (((unsigned char *)alsa_snd_buf)[3*j+1] << 16)
+ | (((unsigned char *)alsa_snd_buf)[3*j+2] << 24)))
+ * (1./ INT32_MAX);
+ }
+#else
+ fprintf(stderr("big endian 24-bit not supported");
+#endif
+ }
else
{
for (i = 0; i < chans; i++, ch++, fp1 += DEFDACBLKSIZE)
diff --git a/src/s_audio_pa.c b/src/s_audio_pa.c
index e02cbc9e1f6ba91f1f7c5bdcc053b4693757aaba..4ceeb6c6aa7423524a7a8a77ea7af2091c1237b9 100644
--- a/src/s_audio_pa.c
+++ b/src/s_audio_pa.c
@@ -11,6 +11,7 @@
#include "s_stuff.h"
#include <stdio.h>
#include <stdlib.h>
+#include <unistd.h>
#include <portaudio.h>
#include "s_audio_pablio.h"
@@ -29,6 +30,34 @@ static t_audiocallback pa_callback;
int pa_foo;
+static void pa_init(void)
+{
+ static int initialized;
+ if (!initialized)
+ {
+ /* Initialize PortAudio */
+ /* for some reason Pa_Initialize(0 closes file descriptor 1.
+ As a workaround, dup it to another number and dup2 it back
+ afterward. */
+ int newfd = dup(1);
+ int err = Pa_Initialize();
+ if (newfd >= 0)
+ {
+ dup2(newfd, 1);
+ close(newfd);
+ }
+ if ( err != paNoError )
+ {
+ fprintf( stderr,
+ "Error number %d occured initializing portaudio\n",
+ err);
+ fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
+ return;
+ }
+ initialized = 1;
+ }
+}
+
static int pa_lowlevel_callback(const void *inputBuffer,
void *outputBuffer, unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *outTime, PaStreamCallbackFlags myflags,
@@ -155,25 +184,11 @@ int pa_open_audio(int inchans, int outchans, int rate, t_sample *soundin,
int indeviceno, int outdeviceno, t_audiocallback callbackfn)
{
PaError err;
- static int initialized;
int j, devno, pa_indev = 0, pa_outdev = 0;
pa_callback = callbackfn;
/* fprintf(stderr, "open callback %d\n", (callbackfn != 0)); */
- if (!initialized)
- {
- /* Initialize PortAudio */
- int err = Pa_Initialize();
- if ( err != paNoError )
- {
- fprintf( stderr,
- "Error number %d occured initializing portaudio\n",
- err);
- fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
- return (1);
- }
- initialized = 1;
- }
+ pa_init();
/* post("in %d out %d rate %d device %d", inchans, outchans, rate, deviceno); */
if (inchans > MAX_PA_CHANS)
{
@@ -349,7 +364,7 @@ void pa_listdevs(void) /* lifted from pa_devs.c in portaudio */
int numDevices;
const PaDeviceInfo *pdi;
PaError err;
- Pa_Initialize();
+ pa_init();
numDevices = Pa_GetDeviceCount();
if( numDevices < 0 )
{
@@ -390,7 +405,7 @@ void pa_getdevs(char *indevlist, int *nindevs,
int i, nin = 0, nout = 0, ndev;
*canmulti = 1; /* one dev each for input and output */
- Pa_Initialize();
+ pa_init();
ndev = Pa_GetDeviceCount();
for (i = 0; i < ndev; i++)
{