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quisk-kc4upr/sound.c

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/*
* Sound modules that do not depend on alsa or portaudio
*/
#include <Python.h>
#include <complex.h>
#include <math.h>
#include <sys/time.h>
#include <time.h>
#ifdef MS_WINDOWS
#include <Winsock2.h>
#else
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#endif
#include "quisk.h"
#include "filter.h"
// Thanks to Franco Spinelli for this fix:
// The H101 hardware using the PCM2904 chip has a one-sample delay between
// channels that must be fixed in software. If you have this problem,
// set channel_delay in your config file. The FIX_H101 #define is obsolete
// but still works. It is equivalent to channel_delay = channel_q.
// The structure sound_dev represents a sound device to open. If portaudio_index
// is -1, it is an ALSA sound device; otherwise it is a portaudio device with that
// index. Portaudio devices have names that start with "portaudio". A device name
// can be the null string, meaning the device should not be opened. The sound_dev
// "handle" is either an alsa handle or a portaudio stream if the stream is open;
// otherwise it is NULL for a closed device.
// Set DEBUG_MIC (in quisk.h) to send the microphone samples to the FFT instead of the radio samples.
// The sample rate and mic sample rate must be 48000. Use -c n2adr/conf4.py.
// 0: Normal operation.
// 1: Send filtered mic output to the FFT.
// 2: Send mic playback to the FFT and to the radio sound playback device "Playback".
// 3: Send unfiltered mic output to the FFT.
#if DEBUG_IO
static int debug_timer = 1; // count up number of samples
#endif
static struct sound_dev Capture, Playback, MicCapture, MicPlayback, DigitalInput, DigitalOutput, RawSamplePlayback;
struct sound_dev quisk_DigitalRx1Output;
// These are arrays of all capture and playback devices, and MUST end with NULL:
static struct sound_dev * CaptureDevices[] = {&Capture, &MicCapture, &DigitalInput, NULL};
static struct sound_dev * PlaybackDevices[] = {&Playback, &MicPlayback, &DigitalOutput, &RawSamplePlayback, &quisk_DigitalRx1Output, NULL};
static SOCKET radio_sound_socket = INVALID_SOCKET; // send radio sound samples to a socket
static SOCKET radio_sound_mic_socket = INVALID_SOCKET; // receive mic samples from a socket
static int radio_sound_nshorts; // number of shorts (two bytes) to send
static int radio_sound_mic_nshorts; // number of shorts (two bytes) to receive
struct sound_conf quisk_sound_state; // Current sound status
struct wav_file {
FILE * fp;
char file_name[QUISK_PATH_SIZE];
int enable;
unsigned long samples;
};
static struct wav_file file_rec_audio, file_rec_samples, file_rec_mic;
static int file_record_button; // the file record button is down
static double digital_output_level = 0.7;
static int dc_remove_bw=100; // bandwidth of DC removal filter
static ty_sample_start pt_sample_start;
static ty_sample_stop pt_sample_stop;
static ty_sample_read pt_sample_read;
ty_sample_write quisk_pt_sample_write;
static complex double cSamples[SAMP_BUFFER_SIZE]; // Complex buffer for samples
#if 0
void quisk_sample_level(const char * msg, complex double * cSamples, int nSamples, double scale)
{
static double time0 = 0;
static double level = 0;
static int count = 0;
double d;
int i;
count += nSamples;
for (i = 0; i < nSamples; i++) {
d = cabs(cSamples[i]);
if (level < d)
level = d;
}
if (QuiskTimeSec() - time0 > 0.1) {
printf ("sample_level %s: %10.6lf count %8d\n", msg, level / scale, count);
level = 0;
count = 0;
time0 = QuiskTimeSec();
}
}
#endif
void ptimer(int counts) // used for debugging
{ // print the number of counts per second
static unsigned int calls=0, total=0;
static time_t time0=0;
time_t dt;
if (time0 == 0) {
time0 = (int)(QuiskTimeSec() * 1.e6);
return;
}
total += counts;
calls++;
if (calls % 1000 == 0) {
dt = (int)(QuiskTimeSec() * 1.e6) - time0;
printf("ptimer: %d counts in %d microseconds %.3f counts/sec\n",
total, (unsigned)dt, (double)total * 1E6 / dt);
}
}
static void delay_sample (struct sound_dev * dev, double * dSamp, int nSamples)
{ // Delay the I or Q data stream by one sample.
// cSamples is double D[nSamples][2]
double d;
double * first, * last;
if (nSamples < 1)
return;
if (dev->channel_Delay == dev->channel_I) {
first = dSamp;
last = dSamp + nSamples * 2 - 2;
}
else if (dev->channel_Delay == dev->channel_Q) {
first = dSamp + 1;
last = dSamp + nSamples * 2 - 1;
}
else {
return;
}
d = dev->save_sample;
dev->save_sample = *last;
while (--nSamples) {
*last = *(last - 2);
last -= 2;
}
*first = d;
}
static void correct_sample (struct sound_dev * dev, complex double * cSamples, int nSamples)
{ // Correct the amplitude and phase
int i;
double re, im;
if (dev->doAmplPhase) { // amplitude and phase corrections
for (i = 0; i < nSamples; i++) {
re = creal(cSamples[i]);
im = cimag(cSamples[i]);
re = re * dev->AmPhAAAA;
im = re * dev->AmPhCCCC + im * dev->AmPhDDDD;
cSamples[i] = re + I * im;
}
}
}
static void DCremove(complex double * cSamples, int nSamples, int sample_rate, int key_state)
{
int i;
double omega, Qsin, Qcos, H0, x;
complex double c;
static int old_sample_rate = 0;
static int old_bandwidth = 0;
static double alpha = 0.95;
static complex double dc_remove = 0;
static complex double dc_average = 0; // Average DC component in samples
static complex double dc_sum = 0;
static int dc_count = 0;
static int dc_key_delay = 0;
if (sample_rate != old_sample_rate || dc_remove_bw != old_bandwidth) {
old_sample_rate = sample_rate; // calculate a new alpha
old_bandwidth = dc_remove_bw;
if (old_bandwidth > 0) {
omega = M_PI * old_bandwidth / (old_sample_rate / 2.0);
Qsin = sin(omega);
Qcos = cos(omega);
H0 = 1.0 / sqrt(2.0);
x = ((Qcos - 1) * (Qcos - 1) + Qsin * Qsin) / (H0 * H0) - Qsin * Qsin;
x = sqrt(x);
alpha = Qcos - x;
//printf ("DC remove: alpha %.3f rate %i bw %i\n", alpha, old_sample_rate, old_bandwidth);
}
else {
//printf("DC remove: disable\n");
}
}
if (quisk_is_vna || old_bandwidth == 0) {
}
else if (old_bandwidth == 1) {
if (key_state) {
dc_key_delay = 0;
dc_sum = 0;
dc_count = 0;
}
else if (dc_key_delay < old_sample_rate) {
dc_key_delay += nSamples;
}
else {
dc_count += nSamples;
for (i = 0; i < nSamples; i++) // Correction for DC offset in samples
dc_sum += cSamples[i];
if (dc_count > old_sample_rate * 2) {
dc_average = dc_sum / dc_count;
//printf("dc average %lf %lf %d\n", creal(dc_average), cimag(dc_average), dc_count);
//printf("dc polar %.0lf %d\n", cabs(dc_average),
// (int)(360.0 / 2 / M_PI * atan2(cimag(dc_average), creal(dc_average))));
dc_sum = 0;
dc_count = 0;
}
}
for (i = 0; i < nSamples; i++) // Correction for DC offset in samples
cSamples[i] -= dc_average;
}
else if (old_bandwidth > 1) {
for (i = 0; i < nSamples; i++) { // DC removal; R.G. Lyons page 553; 3rd Ed. p 762
c = cSamples[i] + dc_remove * alpha;
cSamples[i] = c - dc_remove;
dc_remove = c;
}
}
}
static void record_audio(struct wav_file * wavfile, complex double * cSamples, int nSamples)
{ // Record the speaker audio to a WAV file, PCM, 16 bits, one channel
// TODO: correct for big-endian byte order
FILE * fp;
int j;
short samp; // must be 2 bytes
unsigned int u; // must be 4 bytes
unsigned short s; // must be 2 bytes
switch (nSamples) {
case -1: // Open the file
if (wavfile->fp)
fclose(wavfile->fp);
wavfile->fp = fp = fopen(wavfile->file_name, "wb");
if ( ! fp) {
wavfile->enable = 0;
return;
}
if (fwrite("RIFF", 1, 4, fp) != 4) {
fclose(fp);
wavfile->fp = NULL;
wavfile->enable = 0;
return;
}
// pcm data, 16-bit samples, one channel
u = 36;
fwrite(&u, 4, 1, fp);
fwrite("WAVE", 1, 4, fp);
fwrite("fmt ", 1, 4, fp);
u = 16;
fwrite(&u, 4, 1, fp);
s = 1; // wave_format_pcm
fwrite(&s, 2, 1, fp);
s = 1; // number of channels
fwrite(&s, 2, 1, fp);
u = Playback.sample_rate; // sample rate
fwrite(&u, 4, 1, fp);
u *= 2;
fwrite(&u, 4, 1, fp);
s = 2;
fwrite(&s, 2, 1, fp);
s = 16;
fwrite(&s, 2, 1, fp);
fwrite("data", 1, 4, fp);
u = 0;
fwrite(&u, 4, 1, fp);
wavfile->samples = 0;
break;
case -2: // close the file
if (wavfile->fp)
fclose(wavfile->fp);
wavfile->fp = NULL;
break;
default: // write the sound data to the file
fp = wavfile->fp;
u = (unsigned int)nSamples;
if (wavfile->samples >= 2147483629 - u) { // limit size to 2**32 - 1
wavfile->samples = ~0;
u = ~0;
fseek(fp, 40, SEEK_SET); // seek from the beginning
fwrite(&u, 4, 1, fp);
fseek(fp, 4, SEEK_SET);
fwrite(&u, 4, 1, fp);
}
else {
wavfile->samples += u;
fseek(fp, 40, SEEK_SET);
u = 2 * wavfile->samples;
fwrite(&u, 4, 1, fp);
fseek(fp, 4, SEEK_SET);
u += 36;
fwrite(&u, 4, 1, fp);
}
fseek(fp, 0, SEEK_END); // seek to the end
for (j = 0; j < nSamples; j++) {
samp = (short)(creal(cSamples[j]) / 65536.0);
fwrite(&samp, 2, 1, fp);
}
break;
}
}
static int record_samples(struct wav_file * wavfile, complex double * cSamples, int nSamples)
{ // Record the samples to a WAV file, two float samples I/Q
FILE * fp; // TODO: correct for big-endian byte order
int j;
float samp; // must be 4 bytes
unsigned int u; // must be 4 bytes
unsigned short s; // must be 2 bytes
switch (nSamples) {
case -1: // Open the file
if (wavfile->fp)
fclose(wavfile->fp);
wavfile->fp = fp = fopen(wavfile->file_name, "wb");
if ( ! fp) {
wavfile->enable = 0;
return 0;
}
if (fwrite("RIFF", 1, 4, fp) != 4) {
fclose(fp);
wavfile->fp = NULL;
wavfile->enable = 0;
return 0;
}
// IEEE float data, two channels
u = 36;
fwrite(&u, 4, 1, fp);
fwrite("WAVE", 1, 4, fp);
fwrite("fmt ", 1, 4, fp);
u = 16;
fwrite(&u, 4, 1, fp);
s = 3; // wave_format_ieee_float
fwrite(&s, 2, 1, fp);
s = 2; // number of channels
fwrite(&s, 2, 1, fp);
u = quisk_sound_state.sample_rate; // sample rate
fwrite(&u, 4, 1, fp);
u *= 8;
fwrite(&u, 4, 1, fp);
s = 8;
fwrite(&s, 2, 1, fp);
s = 32;
fwrite(&s, 2, 1, fp);
// Add a LIST chunk of type INFO for further metadata
fwrite("data", 1, 4, fp);
u = 0;
fwrite(&u, 4, 1, fp);
wavfile->samples = 0;
break;
case -2: // close the file
if (wavfile->fp)
fclose(wavfile->fp);
wavfile->fp = NULL;
wavfile->enable = 0;
break;
default: // write the sound data to the file
fp = wavfile->fp;
if ( ! fp)
return 0;
u = (unsigned int)nSamples;
if (wavfile->samples >= 536870907 - u) { // limit size to 2**32 - 1
wavfile->samples = ~0;
u = ~0;
fseek(fp, 40, SEEK_SET); // seek from the beginning
fwrite(&u, 4, 1, fp);
fseek(fp, 4, SEEK_SET); // seek from the beginning
fwrite(&u, 4, 1, fp);
}
else {
wavfile->samples += u;
fseek(fp, 40, SEEK_SET); // seek from the beginning
u = 8 * wavfile->samples;
fwrite(&u, 4, 1, fp);
fseek(fp, 4, SEEK_SET); // seek from the beginning
u += 36 ;
fwrite(&u, 4, 1, fp);
}
fseek(fp, 0, SEEK_END); // seek to the end
for (j = 0; j < nSamples; j++) {
samp = creal(cSamples[j]) / CLIP32;
fwrite(&samp, 4, 1, fp);
samp = cimag(cSamples[j]) / CLIP32;
fwrite(&samp, 4, 1, fp);
}
break;
}
return 1;
}
void quisk_sample_source(ty_sample_start start, ty_sample_stop stop, ty_sample_read read)
{
pt_sample_start = start;
pt_sample_stop = stop;
pt_sample_read = read;
}
void quisk_sample_source4(ty_sample_start start, ty_sample_stop stop, ty_sample_read read, ty_sample_write write)
{
pt_sample_start = start;
pt_sample_stop = stop;
pt_sample_read = read;
quisk_pt_sample_write = write;
}
/*!
* \brief Driver interface for reading samples from a device
*
* \param dev Input. Device to read from
* \param cSamples Output. Read samples.
* \returns number of samples read
*/
int read_sound_interface(
struct sound_dev* dev,
complex double * cSamples
)
{
int i, nSamples;
double avg, samp, re, im, frac, diff;
// Read using correct driver.
switch( dev->driver )
{
case DEV_DRIVER_PORTAUDIO:
#ifdef QUISK_NO_PORTAUDIO
nSamples = 0;
#else
nSamples = quisk_read_portaudio(dev, cSamples);
#endif
break;
case DEV_DRIVER_ALSA:
nSamples = quisk_read_alsa(dev, cSamples);
break;
case DEV_DRIVER_PULSEAUDIO:
nSamples = quisk_read_pulseaudio(dev, cSamples);
break;
case DEV_DRIVER_NONE:
default:
return 0;
}
if ( ! cSamples || nSamples <= 0 || dev->sample_rate <= 0) // cSamples can be NULL
return nSamples;
// Calculate average squared level
avg = dev->average_square;
frac = 1.0 / (0.2 * dev->sample_rate);
for (i = 0; i < nSamples; i++) {
re = creal(cSamples[i]);
im = cimag(cSamples[i]);
samp = re * re + im * im;
diff = samp - avg;
if (diff >= 0)
avg = samp; // set to peak value
else
avg = avg + frac * diff;
}
dev->average_square = avg;
return nSamples;
}
/*!
* \brief Driver interface for playing samples to a device
*
* \param dev Input. Device to play to
* \param nSamples Input. Number of samples to play
* \param cSamples Input. Samples to play
* \param report_latency Input. 1 to report latency, 0 otherwise.
* \param volume Input. [0,1] volume ratio
* \returns number of samples read
*/
void play_sound_interface(
struct sound_dev* dev,
int nSamples,
complex double * cSamples,
int report_latency,
double volume
)
{
int i;
double avg, samp, re, im, frac, diff;
if (cSamples && nSamples > 0 && dev->sample_rate > 0) {
// Calculate average squared level
avg = dev->average_square;
frac = 1.0 / (0.2 * dev->sample_rate);
for (i = 0; i < nSamples; i++) {
re = creal(cSamples[i]);
im = cimag(cSamples[i]);
samp = re * re + im * im;
diff = samp - avg;
if (diff >= 0)
avg = samp; // set to peak value
else
avg = avg + frac * diff;
}
dev->average_square = avg;
}
// Play using correct driver.
switch( dev->driver )
{
case DEV_DRIVER_PORTAUDIO:
#ifndef QUISK_NO_PORTAUDIO
quisk_play_portaudio(dev, nSamples, cSamples, report_latency, volume);
#endif
break;
case DEV_DRIVER_ALSA:
quisk_play_alsa(dev, nSamples, cSamples, report_latency, volume);
break;
case DEV_DRIVER_PULSEAUDIO:
quisk_play_pulseaudio(dev, nSamples, cSamples, report_latency, volume);
break;
case DEV_DRIVER_NONE:
default:
break;
}
}
static int read_radio_sound_socket(complex double * cSamples)
{
int i, bytes, nSamples;
short s;
double d;
struct timeval tm_wait;
char buf[1500];
fd_set fds;
static int started = 0;
nSamples = 0;
while (1) { // read all available blocks
if (nSamples > SAMP_BUFFER_SIZE / 2)
break;
tm_wait.tv_sec = 0;
tm_wait.tv_usec = 0;
FD_ZERO (&fds);
FD_SET (radio_sound_mic_socket, &fds);
if (select (radio_sound_mic_socket + 1, &fds, NULL, NULL, &tm_wait) != 1)
break;
bytes = recv(radio_sound_mic_socket, buf, 1500, 0);
if (bytes == radio_sound_mic_nshorts * 2) { // required block size
started = 1;
for (i = 2; i < bytes; i += 2) {
memcpy(&s, buf + i, 2);
d = (double)s / CLIP16 * CLIP32; // convert 16-bit samples to 32 bits
cSamples[nSamples++] = d + I * d;
}
}
}
if ( ! started && nSamples == 0) {
i = send(radio_sound_mic_socket, "rr", 2, 0);
if (i != 2)
printf("read_radio_sound_mic_socket returned %d\n", i);
}
return nSamples;
}
static void send_radio_sound_socket(complex double * cSamples, int count, double volume)
{ // Send count samples. Each sample is sent as two shorts (4 bytes) of I/Q data.
// Send an initial two bytes of zero for each block.
// Transmission is delayed until a whole block of data is available.
int i, sent;
static short udp_iq[750] = {0}; // Documented maximum radio sound samples is 367
static int udp_size = 1;
for (i = 0; i < count; i++) {
udp_iq[udp_size++] = (short)(creal(cSamples[i]) * volume * (double)CLIP16 / CLIP32);
udp_iq[udp_size++] = (short)(cimag(cSamples[i]) * volume * (double)CLIP16 / CLIP32);
if (udp_size >= radio_sound_nshorts) { // check count
sent = send(radio_sound_socket, (char *)udp_iq, udp_size * 2, 0);
if (sent != udp_size * 2)
printf("Send audio socket returned %d\n", sent);
udp_size = 1;
}
}
}
int quisk_read_sound(void) // Called from sound thread
{ // called in an infinite loop by the main program
int i, nSamples, mic_count, mic_interp, retval, is_cw, mic_sample_rate;
double mic_play_volume;
complex double tx_mic_phase;
static double cwEnvelope=0;
static double cwCount=0;
static complex double tuneVector = (double)CLIP32 / CLIP16; // Convert 16-bit to 32-bit samples
static struct quisk_cFilter filtInterp={NULL};
int key_state, is_DGT;
#if DEBUG_MIC == 1
complex double tmpSamples[SAMP_BUFFER_SIZE];
#endif
quisk_sound_state.interupts++;
key_state = quisk_is_key_down(); //reading this once is important for predicable bevavior on cork/flush
#if DEBUG_IO > 1
QuiskPrintTime("Start read_sound", 0);
#endif
#ifndef MS_WINDOWS
if (quisk_sound_state.IQ_server[0] && ! (rxMode == CWL || rxMode == CWU)) {
if (Capture.handle && Capture.driver == DEV_DRIVER_PULSEAUDIO) {
if (key_state == 1 && !Capture.cork_status)
quisk_cork_pulseaudio(&Capture, 1);
else if (key_state == 0 && Capture.cork_status) {
quisk_cork_pulseaudio(&Capture, 0);
quisk_flush_pulseaudio(&Capture);
}
}
if (MicPlayback.handle && MicPlayback.driver == DEV_DRIVER_PULSEAUDIO) {
if (key_state == 0 && !MicPlayback.cork_status)
quisk_cork_pulseaudio(&MicPlayback, 1);
else if (key_state == 1 && MicPlayback.cork_status) {
quisk_cork_pulseaudio(&MicPlayback, 0);
quisk_flush_pulseaudio(&MicPlayback);
}
}
}
else if (quisk_sound_state.IQ_server[0]) {
if (Capture.handle && Capture.driver == DEV_DRIVER_PULSEAUDIO) {
if (Capture.cork_status)
quisk_cork_pulseaudio(&Capture, 0);
}
if (MicPlayback.handle && MicPlayback.driver == DEV_DRIVER_PULSEAUDIO) {
if (MicPlayback.cork_status)
quisk_cork_pulseaudio(&MicPlayback, 0);
}
}
#endif
if (pt_sample_read) { // read samples from SDR-IQ or UDP or SoapySDR
nSamples = (*pt_sample_read)(cSamples);
DCremove(cSamples, nSamples, quisk_sound_state.sample_rate, key_state);
if (nSamples <= 0)
QuiskSleepMicrosec(2000);
}
else if (Capture.handle) { // blocking read from soundcard
nSamples = read_sound_interface(&Capture, cSamples);
if (Capture.channel_Delay >= 0) // delay the I or Q channel by one sample
delay_sample(&Capture, (double *)cSamples, nSamples);
if (Capture.doAmplPhase) // amplitude and phase corrections
correct_sample(&Capture, cSamples, nSamples);
DCremove(cSamples, nSamples, quisk_sound_state.sample_rate, key_state);
if (nSamples <= 0)
QuiskSleepMicrosec(2000);
}
else {
QuiskSleepMicrosec(5000);
nSamples = QuiskDeltaMsec(1) * quisk_sound_state.sample_rate / 1000;
if (nSamples > SAMP_BUFFER_SIZE / 2)
nSamples = SAMP_BUFFER_SIZE / 2;
for (i = 0; i < nSamples; i++)
cSamples[i] = 0;
}
retval = nSamples; // retval remains the number of samples read
#if DEBUG_IO
debug_timer += nSamples;
if (debug_timer >= quisk_sound_state.sample_rate) // one second
debug_timer = 0;
#endif
#if DEBUG_IO > 2
ptimer (nSamples);
#endif
quisk_sound_state.latencyCapt = nSamples; // samples available
#if DEBUG_IO > 1
QuiskPrintTime(" read samples", 0);
#endif
// Perhaps record the Rx samples to a file
if ( ! key_state && file_rec_samples.fp)
record_samples(&file_rec_samples, cSamples, nSamples);
// Perhaps write samples to a loopback device for use by another program
if (RawSamplePlayback.handle)
play_sound_interface(&RawSamplePlayback, nSamples, cSamples, 0, 1.0);
// Perhaps replace the samples with samples from a file
if (quisk_record_state == PLAY_SAMPLES)
quisk_play_samples(cSamples, nSamples);
#if ! DEBUG_MIC
nSamples = quisk_process_samples(cSamples, nSamples);
#endif
#if DEBUG_IO > 1
QuiskPrintTime(" process samples", 0);
#endif
is_DGT = rxMode == DGT_U || rxMode == DGT_L || rxMode == DGT_IQ || rxMode == DGT_FM;
if (quisk_record_state == PLAYBACK)
quisk_tmp_playback(cSamples, nSamples, 1.0); // replace radio sound
else if (quisk_record_state == PLAY_FILE)
quisk_file_playback(cSamples, nSamples, 1.0); // replace radio sound
// Play the demodulated audio
#if DEBUG_MIC != 2
play_sound_interface(&Playback, nSamples, cSamples, 1, quisk_audioVolume);
#endif
if (radio_sound_socket != INVALID_SOCKET)
send_radio_sound_socket(cSamples, nSamples, quisk_audioVolume);
// Play digital if required
if (is_DGT)
play_sound_interface(&DigitalOutput, nSamples, cSamples, 1, digital_output_level);
// Perhaps record the speaker audio to a file
if ( ! key_state && file_rec_audio.fp)
record_audio(&file_rec_audio, cSamples, nSamples); // Record Rx samples
#if DEBUG_IO > 1
QuiskPrintTime(" play samples", 0);
#endif
// Read and process the microphone
mic_sample_rate = quisk_sound_state.mic_sample_rate;
if (MicCapture.handle)
mic_count = read_sound_interface(&MicCapture, cSamples);
else if (radio_sound_mic_socket != INVALID_SOCKET)
mic_count = read_radio_sound_socket(cSamples);
else { // No mic source; use zero samples
mic_count = QuiskDeltaMsec(0) * mic_sample_rate / 1000;
if (mic_count > SAMP_BUFFER_SIZE / 2)
mic_count = SAMP_BUFFER_SIZE / 2;
for (i = 0; i < mic_count; i++)
cSamples[i] = 0;
}
if (quisk_record_state == PLAYBACK) // Discard previous samples and replace with saved sound
quisk_tmp_microphone(cSamples, mic_count);
else if (quisk_record_state == PLAY_FILE) // Discard previous samples and replace with saved sound
quisk_file_microphone(cSamples, mic_count);
if (DigitalInput.handle) {
if (is_DGT) { // Discard previous mic samples and use digital samples
mic_sample_rate = DigitalInput.sample_rate;
mic_count = read_sound_interface(&DigitalInput, cSamples);
}
else { // Read and discard any digital samples
read_sound_interface(&DigitalInput, NULL);
}
}
else if (is_DGT) { // Use zero-valued samples
for (i = 0; i < mic_count; i++)
cSamples[i] = 0;
}
//quisk_sample_level("read mic or DGT", cSamples, mic_count, CLIP16);
// Perhaps record the microphone audio to the speaker audio file
if (key_state && file_rec_audio.fp)
record_audio(&file_rec_audio, cSamples, mic_count);
// Perhaps record the microphone audio to the microphone audio file
if (file_rec_mic.fp)
record_audio(&file_rec_mic, cSamples, mic_count);
if (mic_count > 0) {
#if DEBUG_IO > 1
QuiskPrintTime(" mic-read", 0);
#endif
#if DEBUG_MIC == 3
quisk_process_samples(cSamples, mic_count);
#endif
// quisk_process_microphone returns samples at the sample rate MIC_OUT_RATE
mic_count = quisk_process_microphone(mic_sample_rate, cSamples, mic_count);
#if DEBUG_MIC == 1
for (i = 0; i < mic_count; i++)
tmpSamples[i] = cSamples[i] * (double)CLIP32 / CLIP16; // convert 16-bit samples to 32 bits
quisk_process_samples(tmpSamples, mic_count);
#endif
#if DEBUG_IO > 1
QuiskPrintTime(" mic-proc", 0);
#endif
}
//quisk_sample_level("quisk_process_microphone", cSamples, mic_count, CLIP16);
// Mic playback without a mic is needed for CW
if (MicPlayback.handle) { // Mic playback: send mic I/Q samples to a sound card
//quisk_sample_level("MicPlayback.handle", cSamples, mic_count, CLIP16);
mic_play_volume = 1.0;
if (rxMode == CWL || rxMode == CWU) { // Transmit CW
is_cw = 1;
}
else {
is_cw = 0;
cwCount = 0;
cwEnvelope = 0.0;
}
tx_mic_phase = cexp(( -I * 2.0 * M_PI * quisk_tx_tune_freq) / MicPlayback.sample_rate);
if (is_cw) { // Transmit CW; use capture device for timing, not microphone
cwCount += (double)retval * MicPlayback.sample_rate / quisk_sound_state.sample_rate;
mic_count = 0;
if (quisk_is_key_down()) {
while (cwCount >= 1.0) {
if (cwEnvelope < 1.0) {
cwEnvelope += 1. / (MicPlayback.sample_rate * 5e-3); // 5 milliseconds
if (cwEnvelope > 1.0)
cwEnvelope = 1.0;
}
if (quiskSpotLevel >= 0)
cSamples[mic_count++] = (CLIP16 - 1) * cwEnvelope * quiskSpotLevel / 1000.0 * tuneVector * quisk_sound_state.mic_out_volume;
else
cSamples[mic_count++] = (CLIP16 - 1) * cwEnvelope * tuneVector * quisk_sound_state.mic_out_volume;
tuneVector *= tx_mic_phase;
cwCount -= 1;
}
}
else { // key is up
while (cwCount >= 1.0) {
if (cwEnvelope > 0.0) {
cwEnvelope -= 1.0 / (MicPlayback.sample_rate * 5e-3); // 5 milliseconds
if (cwEnvelope < 0.0)
cwEnvelope = 0.0;
}
cSamples[mic_count++] = (CLIP16 - 1) * cwEnvelope * tuneVector * quisk_sound_state.mic_out_volume;
tuneVector *= tx_mic_phase;
cwCount -= 1;
}
}
}
else if( ! DEBUG_MIC && ! quisk_is_key_down()) { // Not CW and key up: zero samples
mic_play_volume = 0.0;
for (i = 0; i < mic_count; i++)
cSamples[i] = 0.0;
}
// Perhaps interpolate the mic samples back to the mic play rate
mic_interp = MicPlayback.sample_rate / MIC_OUT_RATE;
if ( ! is_cw && mic_interp > 1) {
if (! filtInterp.dCoefs)
quisk_filt_cInit(&filtInterp, quiskFilt12_19Coefs, sizeof(quiskFilt12_19Coefs)/sizeof(double));
mic_count = quisk_cInterpolate(cSamples, mic_count, &filtInterp, mic_interp);
}
// Tune the samples to frequency and convert 16-bit samples to 32-bits (using tuneVector)
if ( ! is_cw) {
for (i = 0; i < mic_count; i++) {
cSamples[i] = conj(cSamples[i]) * tuneVector * quisk_sound_state.mic_out_volume;
tuneVector *= tx_mic_phase;
}
}
// delay the I or Q channel by one sample
if (MicPlayback.channel_Delay >= 0)
delay_sample(&MicPlayback, (double *)cSamples, mic_count);
// amplitude and phase corrections
if (MicPlayback.doAmplPhase)
correct_sample (&MicPlayback, cSamples, mic_count);
// play mic samples
//quisk_sample_level("play MicPlayback", cSamples, mic_count, CLIP32);
play_sound_interface(&MicPlayback, mic_count, cSamples, 1, mic_play_volume);
#if DEBUG_MIC == 2
play_sound_interface(&Playback, mic_count, cSamples, 1, quisk_audioVolume);
quisk_process_samples(cSamples, mic_count);
#endif
}
#if DEBUG_IO > 1
QuiskPrintTime(" finished", 0);
#endif
// Return negative number for error
return retval;
}
int quisk_get_overrange(void) // Called from GUI thread
{ // Return the overrange (ADC clip) counter, then zero it
int i;
i = quisk_sound_state.overrange + Capture.overrange;
quisk_sound_state.overrange = 0;
Capture.overrange = 0;
return i;
}
void quisk_close_sound(void) // Called from sound thread
{
#ifdef MS_WINDOWS
int cleanup = radio_sound_socket != INVALID_SOCKET || radio_sound_mic_socket != INVALID_SOCKET;
#endif
#ifndef QUISK_NO_PORTAUDIO
quisk_close_sound_portaudio();
#endif
quisk_close_sound_alsa(CaptureDevices, PlaybackDevices);
quisk_close_sound_pulseaudio();
if (pt_sample_stop)
(*pt_sample_stop)();
strncpy (quisk_sound_state.err_msg, CLOSED_TEXT, QUISK_SC_SIZE);
if (radio_sound_socket != INVALID_SOCKET) {
close(radio_sound_socket);
radio_sound_socket = INVALID_SOCKET;
}
if (radio_sound_mic_socket != INVALID_SOCKET) {
shutdown(radio_sound_mic_socket, QUISK_SHUT_RD);
send(radio_sound_mic_socket, "ss", 2, 0);
send(radio_sound_mic_socket, "ss", 2, 0);
QuiskSleepMicrosec(1000000);
close(radio_sound_mic_socket);
radio_sound_mic_socket = INVALID_SOCKET;
}
#ifdef MS_WINDOWS
if (cleanup)
WSACleanup();
#endif
}
static void set_num_channels(struct sound_dev * dev)
{ // Set num_channels to the maximum channel index plus one
dev->num_channels = dev->channel_I;
if (dev->num_channels < dev->channel_Q)
dev->num_channels = dev->channel_Q;
dev->num_channels++;
}
//! \brief Returns 1 if \c string starts with \c prefix. 0 otherwise.
int starts_with( const char* string, const char* prefix )
{
size_t plen = strlen(prefix);
if( strlen(string) < plen )
return 0;
else
return strncmp( string, prefix, plen ) == 0 ? 1 : 0;
}
/*!
* \brief From the sound_dev.name field, decide which driver to use for which device
*/
void decide_drivers(
struct sound_dev** pDevs
)
{
const char* name;
// No name means no driver.
// If name starts with 'portaudio', it's portaudio. Else, if it starts with
// 'pulse', it's PulseAudio. Else, if it starts with 'alsa', it's ALSA.
// Otherwise, just guess ALSA.
while(1)
{
struct sound_dev* dev = *pDevs++;
if( !dev )
break;
name = dev->name;
if( ! name || name[0] == '\0' )
dev->driver = DEV_DRIVER_NONE;
else if( starts_with(name, "portaudio") )
dev->driver = DEV_DRIVER_PORTAUDIO;
else if( starts_with(name, "pulse") )
dev->driver = DEV_DRIVER_PULSEAUDIO;
else if( starts_with(name, "alsa") )
dev->driver = DEV_DRIVER_ALSA;
else
dev->driver = DEV_DRIVER_ALSA;
}
}
static void open_radio_sound_socket(void)
{
struct sockaddr_in Addr;
int samples, port, sndsize = 48000;
char radio_sound_ip[QUISK_SC_SIZE];
char radio_sound_mic_ip[QUISK_SC_SIZE];
#ifdef MS_WINDOWS
WORD wVersionRequested;
WSADATA wsaData;
#endif
dc_remove_bw = QuiskGetConfigInt ("dc_remove_bw", 100);
strncpy(radio_sound_ip, QuiskGetConfigString ("radio_sound_ip", ""), QUISK_SC_SIZE);
strncpy(radio_sound_mic_ip, QuiskGetConfigString ("radio_sound_mic_ip", ""), QUISK_SC_SIZE);
if (radio_sound_ip[0] == 0 && radio_sound_mic_ip[0] == 0)
return;
#ifdef MS_WINDOWS
wVersionRequested = MAKEWORD(2, 2);
if (WSAStartup(wVersionRequested, &wsaData) != 0) {
printf("open_radio_sound_socket: Failure to start WinSock\n");
return; // failure to start winsock
}
#endif
if (radio_sound_ip[0]) {
port = QuiskGetConfigInt ("radio_sound_port", 0);
samples = QuiskGetConfigInt ("radio_sound_nsamples", 360);
if (samples > 367)
samples = 367;
radio_sound_nshorts = samples * 2 + 1;
radio_sound_socket = socket(PF_INET, SOCK_DGRAM, 0);
if (radio_sound_socket != INVALID_SOCKET) {
setsockopt(radio_sound_socket, SOL_SOCKET, SO_SNDBUF, (char *)&sndsize, sizeof(sndsize));
Addr.sin_family = AF_INET;
Addr.sin_port = htons(port);
#ifdef MS_WINDOWS
Addr.sin_addr.S_un.S_addr = inet_addr(radio_sound_ip);
#else
inet_aton(radio_sound_ip, &Addr.sin_addr);
#endif
if (connect(radio_sound_socket, (const struct sockaddr *)&Addr, sizeof(Addr)) != 0) {
close(radio_sound_socket);
radio_sound_socket = INVALID_SOCKET;
}
}
if (radio_sound_socket == INVALID_SOCKET) {
printf("open_radio_sound_socket: Failure to open socket\n");
}
else {
#if DEBUG_IO
printf("open_radio_sound_socket: opened socket %s\n", radio_sound_ip);
#endif
}
}
if (radio_sound_mic_ip[0]) {
port = QuiskGetConfigInt ("radio_sound_mic_port", 0);
samples = QuiskGetConfigInt ("radio_sound_mic_nsamples", 720);
if (samples > 734)
samples = 734;
radio_sound_mic_nshorts = samples + 1;
radio_sound_mic_socket = socket(PF_INET, SOCK_DGRAM, 0);
if (radio_sound_mic_socket != INVALID_SOCKET) {
setsockopt(radio_sound_mic_socket, SOL_SOCKET, SO_SNDBUF, (char *)&sndsize, sizeof(sndsize));
Addr.sin_family = AF_INET;
Addr.sin_port = htons(port);
#ifdef MS_WINDOWS
Addr.sin_addr.S_un.S_addr = inet_addr(radio_sound_mic_ip);
#else
inet_aton(radio_sound_mic_ip, &Addr.sin_addr);
#endif
if (connect(radio_sound_mic_socket, (const struct sockaddr *)&Addr, sizeof(Addr)) != 0) {
close(radio_sound_mic_socket);
radio_sound_mic_socket = INVALID_SOCKET;
}
}
if (radio_sound_mic_socket == INVALID_SOCKET) {
printf("open_radio_sound_mic_socket: Failure to open socket\n");
}
else {
#if DEBUG_IO
printf("open_radio_sound_mic_socket: opened socket %s\n", radio_sound_mic_ip);
#endif
}
}
}
void quisk_open_sound(void) // Called from GUI thread
{
int i;
quisk_sound_state.read_error = 0;
quisk_sound_state.write_error = 0;
quisk_sound_state.underrun_error = 0;
quisk_sound_state.mic_read_error = 0;
quisk_sound_state.interupts = 0;
quisk_sound_state.rate_min = quisk_sound_state.rate_max = -99;
quisk_sound_state.chan_min = quisk_sound_state.chan_max = -99;
quisk_sound_state.msg1[0] = 0;
quisk_sound_state.err_msg[0] = 0;
// Set stream names
strncpy(Capture.name, quisk_sound_state.dev_capt_name, QUISK_SC_SIZE);
strncpy(Playback.name, quisk_sound_state.dev_play_name, QUISK_SC_SIZE);
strncpy(MicCapture.name, quisk_sound_state.mic_dev_name, QUISK_SC_SIZE);
strncpy(MicPlayback.name, quisk_sound_state.name_of_mic_play, QUISK_SC_SIZE);
strncpy(DigitalInput.name, QuiskGetConfigString ("digital_input_name", ""), QUISK_SC_SIZE);
strncpy(DigitalOutput.name, QuiskGetConfigString ("digital_output_name", ""), QUISK_SC_SIZE);
strncpy(RawSamplePlayback.name, QuiskGetConfigString ("sample_playback_name", ""), QUISK_SC_SIZE);
strncpy(quisk_DigitalRx1Output.name, QuiskGetConfigString ("digital_rx1_name", ""), QUISK_SC_SIZE);
// Set stream descriptions. This is important for "deviceless" drivers like
// PulseAudio to be able to distinguish the streams from each other.
strncpy(Capture.stream_description, "I/Q Rx Sample Input", QUISK_SC_SIZE);
Capture.stream_description[QUISK_SC_SIZE-1] = '\0';
strncpy(Playback.stream_description, "Radio Sound Output", QUISK_SC_SIZE);
Playback.stream_description[QUISK_SC_SIZE-1] = '\0';
strncpy(MicCapture.stream_description, "Microphone Input", QUISK_SC_SIZE);
MicCapture.stream_description[QUISK_SC_SIZE-1] = '\0';
strncpy(MicPlayback.stream_description, "I/Q Tx Sample Output", QUISK_SC_SIZE);
MicPlayback.stream_description[QUISK_SC_SIZE-1] = '\0';
strncpy(DigitalInput.stream_description, "External Digital Input", QUISK_SC_SIZE);
strncpy(DigitalOutput.stream_description, "External Digital Output", QUISK_SC_SIZE);
strncpy(RawSamplePlayback.stream_description, "Raw Digital Output", QUISK_SC_SIZE);
strncpy(quisk_DigitalRx1Output.stream_description, "Digital Rx1 Output", QUISK_SC_SIZE);
Playback.sample_rate = quisk_sound_state.playback_rate; // Radio sound play rate
MicPlayback.sample_rate = quisk_sound_state.mic_playback_rate;
MicCapture.sample_rate = quisk_sound_state.mic_sample_rate;
MicCapture.channel_I = quisk_sound_state.mic_channel_I; // Mic audio is here
MicCapture.channel_Q = quisk_sound_state.mic_channel_Q;
// Capture device for digital modes
DigitalInput.sample_rate = 48000;
DigitalInput.channel_I = 0;
DigitalInput.channel_Q = 1;
// Playback device for digital modes
digital_output_level = QuiskGetConfigDouble("digital_output_level", 0.7);
DigitalOutput.sample_rate = quisk_sound_state.playback_rate; // Radio sound play rate
DigitalOutput.channel_I = 0;
DigitalOutput.channel_Q = 1;
// Playback device for raw samples
RawSamplePlayback.sample_rate = quisk_sound_state.sample_rate;
RawSamplePlayback.channel_I = 0;
RawSamplePlayback.channel_Q = 1;
// Playback device for digital modes from sub-receivers
quisk_DigitalRx1Output.sample_rate = 48000;
quisk_DigitalRx1Output.channel_I = 0;
quisk_DigitalRx1Output.channel_Q = 1;
set_num_channels (&Capture);
set_num_channels (&Playback);
set_num_channels (&MicCapture);
set_num_channels (&MicPlayback);
set_num_channels (&DigitalInput);
set_num_channels (&DigitalOutput);
set_num_channels (&RawSamplePlayback);
set_num_channels (&quisk_DigitalRx1Output);
Capture.average_square = 0;
Playback.average_square = 0;
MicCapture.average_square = 0;
MicPlayback.average_square = 0;
DigitalInput.average_square = 0;
DigitalOutput.average_square = 0;
RawSamplePlayback.average_square = 0;
quisk_DigitalRx1Output.average_square = 0;
//Needed for pulse audio context connection (KM4DSJ)
Capture.stream_dir_record = 1;
Playback.stream_dir_record = 0;
MicCapture.stream_dir_record = 1;
MicPlayback.stream_dir_record= 0;
DigitalInput.stream_dir_record = 1;
DigitalOutput.stream_dir_record = 0;
RawSamplePlayback.stream_dir_record = 0;
quisk_DigitalRx1Output.stream_dir_record = 0;
//For remote IQ server over pulseaudio (KM4DSJ)
if (quisk_sound_state.IQ_server[0]) {
strncpy(Capture.server, quisk_sound_state.IQ_server, IP_SIZE);
strncpy(MicPlayback.server, quisk_sound_state.IQ_server, IP_SIZE);
}
#ifdef FIX_H101
Capture.channel_Delay = Capture.channel_Q; // Obsolete; do not use.
#else
Capture.channel_Delay = QuiskGetConfigInt ("channel_delay", -1);
#endif
MicPlayback.channel_Delay = QuiskGetConfigInt ("tx_channel_delay", -1);
if (pt_sample_read) // capture from SDR-IQ by Rf-Space or UDP
Capture.name[0] = 0; // zero the capture soundcard name
else // sound card capture
Capture.sample_rate = quisk_sound_state.sample_rate;
// set read size for sound card capture
i = (int)(quisk_sound_state.data_poll_usec * 1e-6 * Capture.sample_rate + 0.5);
i = i / 64 * 64;
if (i > SAMP_BUFFER_SIZE / Capture.num_channels) // limit to buffer size
i = SAMP_BUFFER_SIZE / Capture.num_channels;
Capture.read_frames = i;
MicCapture.read_frames = 0; // Use non-blocking read for microphone
Playback.read_frames = 0;
MicPlayback.read_frames = 0;
// set sound card play latency
Playback.latency_frames = Playback.sample_rate * quisk_sound_state.latency_millisecs / 1000;
MicPlayback.latency_frames = MicPlayback.sample_rate * quisk_sound_state.latency_millisecs / 1000;
Capture.latency_frames = 0;
MicCapture.latency_frames = 0;
// set capture and playback for digital modes
DigitalInput.read_frames = 0; // Use non-blocking read
DigitalInput.latency_frames = 0;
DigitalOutput.read_frames = 0;
DigitalOutput.latency_frames = DigitalOutput.sample_rate * 500 / 1000; // 500 milliseconds
quisk_DigitalRx1Output.read_frames = 0;
quisk_DigitalRx1Output.latency_frames = quisk_DigitalRx1Output.sample_rate * 500 / 1000; // 500 milliseconds
// set capture and playback for raw samples
RawSamplePlayback.read_frames = 0;
RawSamplePlayback.latency_frames = RawSamplePlayback.sample_rate * 500 / 1000; // 500 milliseconds
open_radio_sound_socket();
#if DEBUG_IO
printf("Sample buffer size %d, latency msec %d\n", SAMP_BUFFER_SIZE, quisk_sound_state.latency_millisecs);
#endif
}
void quisk_start_sound(void) // Called from sound thread
{
if (pt_sample_start)
(*pt_sample_start)();
// Decide which drivers start which devices.
decide_drivers(CaptureDevices);
decide_drivers(PlaybackDevices);
// Let the drivers see the devices and start them up if appropriate
#ifndef QUISK_NO_PORTAUDIO
quisk_start_sound_portaudio(CaptureDevices, PlaybackDevices);
#endif
quisk_start_sound_pulseaudio(CaptureDevices, PlaybackDevices);
quisk_start_sound_alsa(CaptureDevices, PlaybackDevices);
if (pt_sample_read) { // Capture from SDR-IQ or UDP
quisk_sound_state.rate_min = Playback.rate_min;
quisk_sound_state.rate_max = Playback.rate_max;
quisk_sound_state.chan_min = Playback.chan_min;
quisk_sound_state.chan_max = Playback.chan_max;
}
else { // Capture from sound card
quisk_sound_state.rate_min = Capture.rate_min;
quisk_sound_state.rate_max = Capture.rate_max;
quisk_sound_state.chan_min = Capture.chan_min;
quisk_sound_state.chan_max = Capture.chan_max;
}
QuiskDeltaMsec(0); // Set timer to zero
QuiskDeltaMsec(1);
}
PyObject * quisk_set_ampl_phase(PyObject * self, PyObject * args) // Called from GUI thread
{ /* Set the sound card amplitude and phase corrections. See
S.W. Ellingson, Correcting I-Q Imbalance in Direct Conversion Receivers, February 10, 2003 */
struct sound_dev * dev;
double ampl, phase;
int is_tx; // Is this for Tx? Otherwise Rx.
if (!PyArg_ParseTuple (args, "ddi", &ampl, &phase, &is_tx))
return NULL;
if (is_tx)
dev = &MicPlayback;
else
dev = &Capture;
if (ampl == 0.0 && phase == 0.0) {
dev->doAmplPhase = 0;
}
else {
dev->doAmplPhase = 1;
ampl = ampl + 1.0; // Change factor 0.01 to 1.01
phase = (phase / 360.0) * 2.0 * M_PI; // convert to radians
dev->AmPhAAAA = 1.0 / ampl;
dev->AmPhCCCC = - dev->AmPhAAAA * tan(phase);
dev->AmPhDDDD = 1.0 / cos(phase);
}
Py_INCREF (Py_None);
return Py_None;
}
PyObject * quisk_capt_channels(PyObject * self, PyObject * args) // Called from GUI thread
{
if (!PyArg_ParseTuple (args, "ii", &Capture.channel_I, &Capture.channel_Q))
return NULL;
Py_INCREF (Py_None);
return Py_None;
}
PyObject * quisk_play_channels(PyObject * self, PyObject * args) // Called from GUI thread
{
if (!PyArg_ParseTuple (args, "ii", &Playback.channel_I, &Playback.channel_Q))
return NULL;
Py_INCREF (Py_None);
return Py_None;
}
PyObject * quisk_micplay_channels(PyObject * self, PyObject * args) // Called from GUI thread
{
if (!PyArg_ParseTuple (args, "ii", &MicPlayback.channel_I, &MicPlayback.channel_Q))
return NULL;
Py_INCREF (Py_None);
return Py_None;
}
PyObject * quisk_set_sparams(PyObject * self, PyObject * args, PyObject * keywds)
{ /* Call with keyword arguments ONLY; change local parameters */
static char * kwlist[] = {"dc_remove_bw", "digital_output_level", NULL} ;
if (!PyArg_ParseTupleAndKeywords (args, keywds, "|id", kwlist, &dc_remove_bw, &digital_output_level))
return NULL;
Py_INCREF (Py_None);
return Py_None;
}
void quisk_udp_mic_error(char * msg)
{
MicCapture.dev_error++;
#if DEBUG_IO
printf("%s\n", msg);
#endif
}
static void AddCard(struct sound_dev * dev, PyObject * pylist)
{
PyObject * v;
if (dev->name[0]) {
v = Py_BuildValue("(NNiiid)",
PyUnicode_DecodeUTF8(dev->stream_description, strlen(dev->stream_description), "replace"),
PyUnicode_DecodeUTF8(dev->name, strlen(dev->name), "replace"),
dev->sample_rate, dev->dev_latency, dev->dev_error + dev->dev_underrun, dev->average_square);
PyList_Append(pylist, v);
}
}
PyObject * quisk_sound_errors(PyObject * self, PyObject * args)
{ // return a list of strings with card names and error counts
PyObject * pylist;
if (!PyArg_ParseTuple (args, ""))
return NULL;
pylist = PyList_New(0);
AddCard(&Capture, pylist);
AddCard(&MicCapture, pylist);
AddCard(&DigitalInput, pylist);
AddCard(&Playback, pylist);
AddCard(&MicPlayback, pylist);
AddCard(&DigitalOutput, pylist);
AddCard(&RawSamplePlayback, pylist);
AddCard(&quisk_DigitalRx1Output, pylist);
return pylist;
}
PyObject * quisk_set_file_name(PyObject * self, PyObject * args, PyObject * keywds) // called from GUI
{ // Set the names and enable state of the recording and playback files.
int which = -1;
const char * name = NULL;
int enable = -1;
int play_button = -1;
int record_button = -1;
static char * kwlist[] = {"which", "name", "enable", "play_button", "record_button", NULL} ;
if (!PyArg_ParseTupleAndKeywords (args, keywds, "|isiii", kwlist, &which, &name, &enable, &play_button, &record_button))
return NULL;
switch (which) {
case 0: // record audio file
if (name)
strncpy(file_rec_audio.file_name, name, QUISK_PATH_SIZE);
if (enable != -1)
file_rec_audio.enable = enable;
break;
case 1: // record sample file
if (name)
strncpy(file_rec_samples.file_name, name, QUISK_PATH_SIZE);
if (enable != -1)
file_rec_samples.enable = enable;
break;
case 2: // record mic file
if (name)
strncpy(file_rec_mic.file_name, name, QUISK_PATH_SIZE);
if (enable != -1)
file_rec_mic.enable = enable;
break;
case 10: // play audio file
break;
case 11: // play samples file
break;
case 12: // play CQ message file
break;
}
if (record_button != -1)
file_record_button = record_button;
if (file_rec_audio.enable && file_record_button){ // Open and Close Rx audio file
if ( ! file_rec_audio.fp)
record_audio(&file_rec_audio, NULL, -1); // Open file
}
else if (file_rec_audio.fp) {
record_audio(&file_rec_audio, NULL, -2); // Close file
}
if (file_rec_mic.enable && file_record_button){ // Open and Close microphone audio file
if ( ! file_rec_mic.fp)
record_audio(&file_rec_mic, NULL, -1);
}
else if (file_rec_mic.fp) {
record_audio(&file_rec_mic, NULL, -2);
}
if (file_rec_samples.enable && file_record_button){ // Open and Close I/Q samples file
if ( ! file_rec_samples.fp)
record_samples(&file_rec_samples, NULL, -1);
}
else if (file_rec_samples.fp) {
record_samples(&file_rec_samples, NULL, -2);
}
Py_INCREF (Py_None);
return Py_None;
}
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