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Upstream version 3.0preH

pull/2/head
Stelios Bounanos 2008-06-26 01:13:52 +01:00
rodzic a684c41611
commit 38c152ab14
6 zmienionych plików z 109 dodań i 99 usunięć
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2
ChangeLog
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@ -91,6 +91,8 @@ Change Log:
42) Increased average power level in MT-63 mode.
43) Significantly decreased cpu usage with changes to qrunner and complex
classes.
44) Forced audio-in channel on PortAudio and PulseAudio to 1 channel (left)
to accommodate Vista MME interface.
2.10.3)
1) Corrected memory leak bug.

26
configure.ac
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@ -9,7 +9,7 @@ dnl major and minor must be integers; patch may
dnl contain other characters or be empty
m4_define(FLDIGI_MAJOR, [3])
m4_define(FLDIGI_MINOR, [0])
m4_define(FLDIGI_PATCH, [preG])
m4_define(FLDIGI_PATCH, [preH])
AC_INIT([fldigi], FLDIGI_MAJOR.FLDIGI_MINOR[FLDIGI_PATCH], [w1hkj AT w1hkj DOT com])
@ -229,16 +229,20 @@ AC_OUTPUT
AC_MSG_RESULT([
Configuration summary:
Version ............................. $VERSION
Version ..................... $VERSION
Static linking ...................... $ac_cv_static
CPU optimizations ................... $ac_cv_opt
Debugging ........................... $ac_cv_debug
Static linking .............. $ac_cv_static
CPU optimizations ........... $ac_cv_opt
Debugging ................... $ac_cv_debug
sndfile ............................. $ac_cv_sndfile
OSS ................................. $ac_cv_oss
PortAudio ........................... $ac_cv_portaudio
PulseAudio .......................... $ac_cv_pulseaudio
hamlib .............................. $ac_cv_hamlib
xml-rpc ............................. $ac_cv_xmlrpc
png ......................... $ac_cv_libpng
jpeg ........................ $ac_cv_libjpeg
sndfile ..................... $ac_cv_sndfile
oss ......................... $ac_cv_oss
portaudio ................... $ac_cv_portaudio
pulseaudio .................. $ac_cv_pulseaudio
hamlib ...................... $ac_cv_hamlib
xmlrpc ...................... $ac_cv_xmlrpc
])

9
src/Makefile.am
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@ -349,9 +349,7 @@ EXTRA_fldigi_SOURCES += \
include/winserial.h \
sockets/Socket.cxx \
sockets/ClientSocket.cxx \
sockets/ServerSocket.cxx \
misc/parseUALR.cxx \
misc/parseQRZ.cxx
sockets/ServerSocket.cxx
# Additional non-source files that we distribute
EXTRA_DIST = \
@ -365,4 +363,7 @@ EXTRA_DIST = \
$(srcdir)/../data/mac/Info.plist.in \
$(srcdir)/../data/win32/fldigi.ico \
dialogs/confdialog.fl \
rigcontrol/rigdialog.fl
rigcontrol/rigdialog.fl \
misc/parseUALR.cxx \
misc/parseQRZ.cxx \
misc/ualr-telnet.pl

9
src/misc/macros.cxx
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@ -375,7 +375,7 @@ void set_env(void)
{ "FLDIGI_LOG_QTH", inpQth->value() },
{ "FLDIGI_LOG_LOCATOR", inpLoc->value() },
{ "FLDIGI_LOG_NOTES", inpNotes->value() },
{ "FLDIGI_AZ", "" }
{ "FLDIGI_AZ", inpAZ->value() }
};
// PATH
@ -416,13 +416,6 @@ void set_env(void)
: -active_modem->get_freq()));
env[FLDIGI_FREQUENCY].val = freq;
// azimuth
int az_int;
char az_str[4];
if (sscanf(inpAZ->value(), "%03d", &az_int) == 1) {
snprintf(az_str, sizeof(az_str), "%d", az_int);
env[FLDIGI_AZ].val = az_str;
}
// debugging vars
#ifndef NDEBUG
unsetenv("FLDIGI_NO_EXEC");

131
src/soundcard/sound.cxx
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@ -47,8 +47,6 @@
#include <semaphore.h>
#include <limits.h>
//#include <bits/stream_iterator.h>
#if USE_OSS
# include <sys/soundcard.h>
#endif
@ -66,12 +64,13 @@
#include "timeops.h"
#include "ringbuffer.h"
// We always read and write 2 channels from/to the audio device.
// * input: we ignore the right channel of captured samples
// * output: we copy the left channel to the right channel,
// unless we are in CW or RTTY mode with QSK or PseudoFSK enabled --
// this data then goes into the right channel.
#define CHANNELS 2
// Define these constants here to avoid littering the code with magic numbers.
// Audio input is mono in SoundPort and SoundPulse, stereo in SoundOSS
#define INPUT_CHANNELS 1
// Audio output is always stereo: we copy the left channel to the right channel,
// unless we are in CW or RTTY mode with QSK or PseudoFSK enabled -- this data
// then goes into the right channel.
#define OUTPUT_CHANNELS 2
// We never write duplicate/QSK/PseudoFSK data to the sound files
#define SNDFILE_CHANNELS 1
@ -756,18 +755,16 @@ SoundPort::SoundPort(const char *in_dev, const char *out_dev)
}
try {
snd_buffer = new float[2 * SND_BUF_LEN];
src_buffer = new float[2 * SND_BUF_LEN];
fbuf = new float[2 * SND_BUF_LEN];
src_buffer = new float[OUTPUT_CHANNELS * SND_BUF_LEN];
fbuf = new float[OUTPUT_CHANNELS * SND_BUF_LEN];
}
catch (const std::bad_alloc& e) {
cerr << "Cannot allocate libsamplerate buffers\n";
throw;
}
memset(snd_buffer, 0, CHANNELS * SND_BUF_LEN);
memset(src_buffer, 0, CHANNELS * SND_BUF_LEN);
memset(fbuf, 0, CHANNELS * SND_BUF_LEN);
memset(src_buffer, 0, OUTPUT_CHANNELS * SND_BUF_LEN);
memset(fbuf, 0, OUTPUT_CHANNELS * SND_BUF_LEN);
}
SoundPort::~SoundPort()
@ -836,7 +833,7 @@ void SoundPort::pause_stream(unsigned dir)
while (sem_trywait(sd[dir].csem) == 0);
sd[dir].state = spa_pause;
if (sem_timedwaitr(sd[dir].csem, 2) == -1 && errno == ETIMEDOUT)
if (sem_timedwaitr(sd[dir].csem, 5) == -1 && errno == ETIMEDOUT)
cerr << __func__ << ": stream " << dir << " wedged\n";
}
@ -858,7 +855,7 @@ void SoundPort::Close(unsigned dir)
// first wait for buffers to be drained and for the
// stop callback to signal us that the stream has
// been stopped
if (sem_timedwaitr(sd[i].csem, 2) == -1 && errno == ETIMEDOUT)
if (sem_timedwaitr(sd[i].csem, 5) == -1 && errno == ETIMEDOUT)
cerr << __func__ << ": stream " << i << " wedged\n";
sd[i].state = spa_continue;
@ -931,12 +928,12 @@ size_t SoundPort::Read(double *buf, size_t count)
src_set_ratio(rx_src_state, sd[0].src_ratio);
}
size_t maxframes = (size_t)floor((sd[0].rb->length() / CHANNELS) * sd[0].src_ratio);
size_t maxframes = (size_t)floor(sd[0].rb->length() * sd[0].src_ratio);
if (unlikely(count > maxframes)) {
size_t n = 0;
while (count > maxframes) {
n += Read(buf, maxframes);
buf += CHANNELS * maxframes;
buf += maxframes;
count -= maxframes;
}
if (count > 0)
@ -950,34 +947,34 @@ size_t SoundPort::Read(double *buf, size_t count)
size_t n = 0;
sd[0].blocksize = SCBLOCKSIZE;
while (n < count) {
if ((r = src_callback_read(rx_src_state, sd[0].src_ratio, count - n, rbuf + n*CHANNELS)) == 0)
if ((r = src_callback_read(rx_src_state, sd[0].src_ratio, count - n, rbuf + n)) == 0)
return n;
n += r;
}
}
else {
bool timeout = false;
WAIT_FOR_COND( (sd[0].rb->read_space() >= count * CHANNELS / sd[0].src_ratio), sd[0].rwsem,
(MAX(1.0, 2 * CHANNELS * count / sd->dev_sample_rate)) );
WAIT_FOR_COND( (sd[0].rb->read_space() >= count / sd[0].src_ratio), sd[0].rwsem,
(MAX(1.0, 2 * count / sd->dev_sample_rate)) );
if (timeout)
throw SndException(ETIMEDOUT);
ringbuffer<float>::vector_type vec[2];
sd[0].rb->get_rv(vec);
if (vec[0].len >= count * CHANNELS) {
if (vec[0].len >= count) {
rbuf = vec[0].buf;
sd[0].advance = vec[0].len;
}
else
sd[0].rb->read(fbuf, count * CHANNELS);
sd[0].rb->read(fbuf, count);
}
if (sd[0].advance) {
sd[0].rb->read_advance(sd[0].advance);
sd[0].advance = 0;
}
// deinterleave first channel into buf
// convert to double
for (size_t i = 0; i < count; i++)
buf[i] = rbuf[CHANNELS * i];
buf[i] = rbuf[i];
#if USE_SNDFILE
if (capture)
@ -996,7 +993,7 @@ size_t SoundPort::Write(double *buf, size_t count)
// copy input to both channels
for (size_t i = 0; i < count; i++)
fbuf[CHANNELS * i] = fbuf[CHANNELS * i + 1] = buf[i];
fbuf[OUTPUT_CHANNELS * i] = fbuf[OUTPUT_CHANNELS * i + 1] = buf[i];
return resample_write(fbuf, count);
}
@ -1010,8 +1007,8 @@ size_t SoundPort::Write_stereo(double *bufleft, double *bufright, size_t count)
// interleave into fbuf
for (size_t i = 0; i < count; i++) {
fbuf[CHANNELS * i] = bufleft[i];
fbuf[CHANNELS * i + 1] = bufright[i];
fbuf[OUTPUT_CHANNELS * i] = bufleft[i];
fbuf[OUTPUT_CHANNELS * i + 1] = bufright[i];
}
return resample_write(fbuf, count);
@ -1020,14 +1017,14 @@ size_t SoundPort::Write_stereo(double *bufleft, double *bufright, size_t count)
size_t SoundPort::resample_write(float* buf, size_t count)
{
size_t maxframes = (size_t)floor((sd[1].rb->length() / CHANNELS) / tx_src_data->src_ratio);
maxframes /= 2;
size_t maxframes = (size_t)floor((sd[1].rb->length() / OUTPUT_CHANNELS) / tx_src_data->src_ratio);
maxframes /= 2; // don't fill the buffer
if (unlikely(count > maxframes)) {
size_t n = 0;
while (count > maxframes) {
n += resample_write(buf, maxframes);
buf += CHANNELS * maxframes;
buf += OUTPUT_CHANNELS * maxframes;
count -= maxframes;
}
if (count > 0)
@ -1035,13 +1032,13 @@ size_t SoundPort::resample_write(float* buf, size_t count)
return n;
}
assert(count * CHANNELS * tx_src_data->src_ratio <= sd[1].rb->length());
assert(count * OUTPUT_CHANNELS * tx_src_data->src_ratio <= sd[1].rb->length());
ringbuffer<float>::vector_type vec[2];
sd[1].rb->get_wv(vec);
float* wbuf = buf;
if (req_sample_rate != sd[1].dev_sample_rate || progdefaults.TX_corr != 0) {
if (vec[0].len >= CHANNELS * (size_t)ceil(count * tx_src_data->src_ratio))
if (vec[0].len >= OUTPUT_CHANNELS * (size_t)ceil(count * tx_src_data->src_ratio))
wbuf = vec[0].buf; // direct write in the rb
else
wbuf = src_buffer;
@ -1062,7 +1059,7 @@ size_t SoundPort::resample_write(float* buf, size_t count)
count = tx_src_data->output_frames_gen;
if (wbuf == vec[0].buf) { // advance write pointer and return
sd[1].rb->write_advance(CHANNELS * count);
sd[1].rb->write_advance(OUTPUT_CHANNELS * count);
sem_trywait(sd[1].rwsem);
return count;
}
@ -1071,11 +1068,11 @@ size_t SoundPort::resample_write(float* buf, size_t count)
// if we didn't do a direct resample into the rb, or didn't resample at all,
// we must now copy buf into the ringbuffer, possibly waiting for space first
bool timeout = false;
WAIT_FOR_COND( (sd[1].rb->write_space() >= CHANNELS * count), sd[1].rwsem,
(MAX(1.0, 2 * CHANNELS * count / sd[1].dev_sample_rate)) );
WAIT_FOR_COND( (sd[1].rb->write_space() >= OUTPUT_CHANNELS * count), sd[1].rwsem,
(MAX(1.0, 2 * OUTPUT_CHANNELS * count / sd[1].dev_sample_rate)) );
if (timeout)
throw SndException(ETIMEDOUT);
sd[1].rb->write(wbuf, CHANNELS * count);
sd[1].rb->write(wbuf, OUTPUT_CHANNELS * count);
return count;
}
@ -1095,7 +1092,7 @@ void SoundPort::flush(unsigned dir)
continue;
sd[i].state = spa_drain;
while (sem_trywait(sd[i].csem) == 0);
if (sem_timedwaitr(sd[i].csem, 2) == -1 && errno == ETIMEDOUT)
if (sem_timedwaitr(sd[i].csem, 5) == -1 && errno == ETIMEDOUT)
cerr << "timeout while flushing stream " << i << endl;
sd[i].state = spa_continue;
}
@ -1110,7 +1107,7 @@ void SoundPort::src_data_reset(unsigned dir)
if (rx_src_state)
src_delete(rx_src_state);
rx_src_state = src_callback_new(src_read_cb, progdefaults.sample_converter,
CHANNELS, &err, &sd[0]);
INPUT_CHANNELS, &err, &sd[0]);
if (!rx_src_state)
throw SndException(src_strerror(err));
sd[0].src_ratio = req_sample_rate / (sd[0].dev_sample_rate * (1.0 + rxppm / 1e6));
@ -1118,13 +1115,14 @@ void SoundPort::src_data_reset(unsigned dir)
else if (dir == 1) {
if (tx_src_state)
src_delete(tx_src_state);
tx_src_state = src_new(progdefaults.sample_converter, CHANNELS, &err);
tx_src_state = src_new(progdefaults.sample_converter, OUTPUT_CHANNELS, &err);
if (!tx_src_state)
throw SndException(src_strerror(err));
tx_src_data->src_ratio = sd[1].dev_sample_rate * (1.0 + txppm / 1e6) / req_sample_rate;
}
rbsize = ceil2((unsigned)(2 * CHANNELS * SCBLOCKSIZE *
int channels[2] = { INPUT_CHANNELS, OUTPUT_CHANNELS };
rbsize = ceil2((unsigned)(2 * channels[dir] * SCBLOCKSIZE *
MAX(req_sample_rate, sd[dir].dev_sample_rate) /
MIN(req_sample_rate, sd[dir].dev_sample_rate)));
if (dir == 0) {
@ -1159,8 +1157,8 @@ long SoundPort::src_read_cb(void* arg, float** data)
// wait for data
bool timeout = false;
WAIT_FOR_COND( (sd->rb->read_space() >= CHANNELS * SCBLOCKSIZE), sd->rwsem,
(MAX(1.0, 2 * CHANNELS * SCBLOCKSIZE / sd->dev_sample_rate)) );
WAIT_FOR_COND( (sd->rb->read_space() >= SCBLOCKSIZE), sd->rwsem,
(MAX(1.0, 2 * SCBLOCKSIZE / sd->dev_sample_rate)) );
if (timeout) {
*data = 0;
return 0;
@ -1172,7 +1170,7 @@ long SoundPort::src_read_cb(void* arg, float** data)
*data = vec[0].buf;
sd->advance = vec[0].len;
return vec[0].len / CHANNELS;
return vec[0].len;
}
void SoundPort::init_stream(unsigned dir)
@ -1210,14 +1208,14 @@ void SoundPort::init_stream(unsigned dir)
if (dir == 0) {
sd[0].params.device = idx;
sd[0].params.channelCount = CHANNELS;
sd[0].params.channelCount = INPUT_CHANNELS;
sd[0].params.sampleFormat = paFloat32;
sd[0].params.suggestedLatency = (*sd[dir].idev)->defaultHighInputLatency;
sd[0].params.hostApiSpecificStreamInfo = NULL;
}
else {
sd[1].params.device = idx;
sd[1].params.channelCount = CHANNELS;
sd[1].params.channelCount = OUTPUT_CHANNELS;
sd[1].params.sampleFormat = paFloat32;
if (Pa_GetHostApiInfo((*sd[dir].idev)->hostApi)->type == paMME)
sd[1].params.suggestedLatency = (*sd[dir].idev)->defaultLowOutputLatency;
@ -1325,7 +1323,7 @@ int SoundPort::stream_process(const void* in, void* out, unsigned long nframes,
if (in) {
switch (sd->state) {
case spa_continue: // write into the rb, post rwsem if we wrote anything
if (sd->rb->write(reinterpret_cast<const float*>(in), CHANNELS * nframes))
if (sd->rb->write(reinterpret_cast<const float*>(in), nframes))
sem_post(sd->rwsem);
break;
case spa_drain: case spa_pause: // post csem once
@ -1338,8 +1336,8 @@ int SoundPort::stream_process(const void* in, void* out, unsigned long nframes,
else if (out) {
float* outf = reinterpret_cast<float*>(out);
// if we are paused just pretend that the rb was empty
size_t nread = (sd->state == spa_pause) ? 0 : sd->rb->read(outf, CHANNELS * nframes);
memset(outf + nread, 0, (CHANNELS * nframes - nread) * sizeof(float)); // fill rest with zeroes
size_t nread = (sd->state == spa_pause) ? 0 : sd->rb->read(outf, OUTPUT_CHANNELS * nframes);
memset(outf + nread, 0, (OUTPUT_CHANNELS * nframes - nread) * sizeof(float)); // fill rest with zeroes
switch (sd->state) {
case spa_continue: // post rwsem if we read anything
@ -1424,8 +1422,8 @@ void SoundPort::probe_supported_rates(const device_iterator& idev)
{
PaStreamParameters params[2];
params[0].device = params[1].device = idev - devs.begin();
params[0].channelCount = (*idev)->maxInputChannels;
params[1].channelCount = (*idev)->maxOutputChannels;
params[0].channelCount = INPUT_CHANNELS;
params[1].channelCount = OUTPUT_CHANNELS;
params[0].sampleFormat = params[1].sampleFormat = paFloat32;
params[0].suggestedLatency = (*idev)->defaultHighInputLatency;
params[1].suggestedLatency = (*idev)->defaultHighOutputLatency;
@ -1469,7 +1467,7 @@ void SoundPort::pa_perror(int err, const char* str)
void SoundPort::init_hostapi_ext(void)
{
#if HAVE_DLOPEN
#if HAVE_DLOPEN && !defined(__CYGWIN__)
void* handle = dlopen(NULL, RTLD_LAZY);
if (!handle)
return;
@ -1479,6 +1477,10 @@ void SoundPort::init_hostapi_ext(void)
set_jack_client_name = (PaError (*)(const char*))dlsym(handle, "PaJack_SetClientName");
if (!(err = dlerror()))
set_jack_client_name(PACKAGE_TARNAME);
# ifndef NDEBUG
else
cerr << "dlsym(PaJack_SetClientName) error: " << err << '\n';
# endif
#endif
}
@ -1493,7 +1495,8 @@ SoundPulse::SoundPulse(const char *dev)
sd[0].stream = sd[1].stream = 0;
sd[0].dir = PA_STREAM_RECORD; sd[1].dir = PA_STREAM_PLAYBACK;
sd[0].stream_params.format = sd[1].stream_params.format = PA_SAMPLE_FLOAT32LE;
sd[0].stream_params.channels = sd[1].stream_params.channels = CHANNELS;
sd[0].stream_params.channels = INPUT_CHANNELS;
sd[1].stream_params.channels = OUTPUT_CHANNELS;
try {
tx_src_data = new SRC_DATA;
@ -1504,9 +1507,9 @@ SoundPulse::SoundPulse(const char *dev)
}
try {
snd_buffer = new float[CHANNELS * SND_BUF_LEN];
src_buffer = new float[CHANNELS * SND_BUF_LEN];
fbuf = new float[CHANNELS * SND_BUF_LEN];
snd_buffer = new float[INPUT_CHANNELS * SND_BUF_LEN];
src_buffer = new float[OUTPUT_CHANNELS * SND_BUF_LEN];
fbuf = new float[MAX(INPUT_CHANNELS, OUTPUT_CHANNELS) * SND_BUF_LEN];
}
catch (const std::bad_alloc& e) {
cerr << "Cannot allocate libsamplerate buffers\n";
@ -1611,7 +1614,7 @@ size_t SoundPulse::Write(double* buf, size_t count)
#endif
for (size_t i = 0; i < count; i++)
fbuf[CHANNELS * i] = fbuf[CHANNELS * i + 1] = buf[i];
fbuf[OUTPUT_CHANNELS * i] = fbuf[OUTPUT_CHANNELS * i + 1] = buf[i];
return resample_write(fbuf, count);
}
@ -1624,8 +1627,8 @@ size_t SoundPulse::Write_stereo(double* bufleft, double* bufright, size_t count)
#endif
for (size_t i = 0; i < count; i++) {
fbuf[CHANNELS * i] = bufleft[i];
fbuf[CHANNELS * i + 1] = bufright[i];
fbuf[OUTPUT_CHANNELS * i] = bufleft[i];
fbuf[OUTPUT_CHANNELS * i + 1] = bufright[i];
}
return resample_write(fbuf, count);
@ -1654,7 +1657,7 @@ size_t SoundPulse::resample_write(float* buf, size_t count)
count = tx_src_data->output_frames_gen;
}
if (pa_simple_write(sd[1].stream, wbuf, count * CHANNELS * sizeof(float), &err) == -1)
if (pa_simple_write(sd[1].stream, wbuf, count * OUTPUT_CHANNELS * sizeof(float), &err) == -1)
throw SndPulseException(err);
return count;
@ -1665,7 +1668,7 @@ long SoundPulse::src_read_cb(void* arg, float** data)
SoundPulse* p = reinterpret_cast<SoundPulse*>(arg);
int err;
if (pa_simple_read(p->sd[0].stream, p->snd_buffer, CHANNELS * sizeof(float) * p->sd[0].blocksize, &err) == -1) {
if (pa_simple_read(p->sd[0].stream, p->snd_buffer, sizeof(float) * p->sd[0].blocksize, &err) == -1) {
cerr << "SoundPulse::pa_simple_read error: " << pa_strerror(err) << '\n';
*data = 0;
return 0;
@ -1699,19 +1702,19 @@ size_t SoundPulse::Read(double *buf, size_t count)
size_t n = 0;
sd[0].blocksize = SCBLOCKSIZE;
while (n < count) {
if ((r = src_callback_read(rx_src_state, sd[0].src_ratio, count - n, fbuf + n*CHANNELS)) == 0)
if ((r = src_callback_read(rx_src_state, sd[0].src_ratio, count - n, fbuf + n)) == 0)
return n;
n += r;
}
}
else {
int err;
if (pa_simple_read(sd[0].stream, fbuf, CHANNELS * sizeof(float) * count, &err) == -1)
if (pa_simple_read(sd[0].stream, fbuf, sizeof(float) * count, &err) == -1)
throw SndPulseException(err);
}
for (size_t i = 0; i < count; i++)
buf[i] = fbuf[CHANNELS * i];
buf[i] = fbuf[i];
#if USE_SNDFILE
if (capture)

31
src/soundcard/soundconf.cxx
Wyświetl plik

@ -157,32 +157,39 @@ static void init_portaudio(void)
// select the correct menu items
const Fl_Menu_Item* menu;
int size;
int idx;
idx = -1;
menu = menuPortInDev->menu();
size = menuPortInDev->size();
for (int i = 0; i < size - 1; i++, menu++) {
if (menu->label() && progdefaults.PortInDevice == menu->label()) {
if (progdefaults.PortInIndex != -1 &&
reinterpret_cast<intptr_t>(menu->user_data()) != progdefaults.PortInIndex)
continue;
menuPortInDev->value(i);
menuPortInDev->set_changed();
break;
idx = i; // near match
if (reinterpret_cast<intptr_t>(menu->user_data()) == progdefaults.PortInIndex ||
progdefaults.PortInIndex == -1) // exact match, or index was never saved
break;
}
}
if (idx >= 0) {
menuPortInDev->value(idx);
menuPortInDev->set_changed();
}
idx = -1;
menu = menuPortOutDev->menu();
size = menuPortOutDev->size();
for (int i = 0; i < size - 1; i++, menu++) {
if (menu->label() && progdefaults.PortOutDevice == menu->label()) {
if (progdefaults.PortOutIndex != -1 &&
reinterpret_cast<intptr_t>(menu->user_data()) != progdefaults.PortOutIndex)
continue;
menuPortOutDev->value(i);
menuPortOutDev->set_changed();
break;
idx = i;
if (reinterpret_cast<intptr_t>(menu->user_data()) == progdefaults.PortOutIndex ||
progdefaults.PortOutIndex == -1)
break;
}
}
if (idx >= 0) {
menuPortOutDev->value(idx);
menuPortOutDev->set_changed();
}
#endif
}