void sheepshaver_state::record_audio(void)
{
static uint8 silence[8192] = {0};
record_audio(silence);
}
static int
stream_callback (const void *input_buffer, void *output_buffer, unsigned long frames_per_buffer, const PaStreamCallbackTimeInfo * time_info, PaStreamCallbackFlags status_flags, void *user_data)
{
float **buffers = (float **) output_buffer;
#ifdef _HAVE_RUBBERBAND_
static gboolean initialized = FALSE;
if (!initialized) {
rubberband_set_max_process_size(rubberband, frames_per_buffer);
initialized = TRUE;
}
#endif
size_t i;
for (i = 0; i < 2; ++i)
{
memset (buffers[i], 0, frames_per_buffer * sizeof (float));
}
if (!ready)
return paContinue;
#ifdef _HAVE_FLUIDSYNTH_
if (reset_audio)
{
fluidsynth_all_notes_off ();
reset_synth_channels ();
reset_audio = FALSE;
return paContinue;
}
unsigned char event_data[MAX_MESSAGE_LENGTH]; //needs to be long enough for variable length messages...
size_t event_length = MAX_MESSAGE_LENGTH;
double event_time;
double until_time = nframes_to_seconds (playback_frame + frames_per_buffer);
#ifdef _HAVE_RUBBERBAND_
gint available = rubberband_available(rubberband);
if((!rubberband_active) || (available < (gint)frames_per_buffer)) {
#endif
while (read_event_from_queue (AUDIO_BACKEND, event_data, &event_length, &event_time, until_time/slowdown))
{//g_print("%x %x %x\n", event_data[0], event_data[1], event_data[2] );
fluidsynth_feed_midi (event_data, event_length); //in fluid.c note fluidsynth api ues fluid_synth_xxx these naming conventions are a bit too similar
}
fluidsynth_render_audio (frames_per_buffer, buffers[0], buffers[1]); //in fluid.c calls fluid_synth_write_float()
// Now get any audio to mix - dump it in the left hand channel for now
event_length = frames_per_buffer;
read_event_from_mixer_queue (AUDIO_BACKEND, (void *) buffers[1], &event_length);
#ifdef _HAVE_RUBBERBAND_
}
//if there is stuff available use it and give buffers[] to rubber band to process
if(rubberband_active)
{
if(available < (gint)frames_per_buffer)
rubberband_process(rubberband, (const float * const*)buffers, frames_per_buffer, 0);
available = rubberband_available(rubberband);
if(available >= (gint)frames_per_buffer)
{
rubberband_retrieve(rubberband, buffers, frames_per_buffer);//re-use buffers[] as they are available...
write_samples_to_rubberband_queue (AUDIO_BACKEND, buffers[0], frames_per_buffer);
write_samples_to_rubberband_queue (AUDIO_BACKEND, buffers[1], frames_per_buffer);
available -= frames_per_buffer;
}
event_length = frames_per_buffer;
read_event_from_rubberband_queue (AUDIO_BACKEND, (unsigned char *) buffers[0], &event_length);
event_length = frames_per_buffer;
read_event_from_rubberband_queue (AUDIO_BACKEND, (unsigned char *) buffers[1], &event_length);
}
#endif //_HAVE_RUBBERBAND_
if (until_time < get_playuntil ())
{
#endif //_HAVE_FLUIDSYNTH_
playback_frame += frames_per_buffer;
update_playback_time (TIMEBASE_PRIO_AUDIO, nframes_to_seconds (playback_frame));
#ifdef _HAVE_FLUIDSYNTH_
}
#endif //_HAVE_FLUIDSYNTH_
// This is probably a bad idea to do heavy work in an audio callback
record_audio(buffers, frames_per_buffer);
return paContinue;
}
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;
complex tx_mic_phase;
static double cwEnvelope=0;
static double cwCount=0;
static complex tuneVector = (double)CLIP32 / CLIP16; // Convert 16-bit to 32-bit samples
static struct quisk_cFilter filtInterp={NULL};
quisk_sound_state.interupts++;
#if DEBUG_IO > 1
QuiskPrintTime("Start read_sound", 0);
#endif
if (pt_sample_read) { // read samples from SDR-IQ
nSamples = (*pt_sample_read)(cSamples);
}
else if (quisk_using_udp) { // read samples from UDP port
nSamples = quisk_read_rx_udp(cSamples);
}
else if (Capture.handle) { // blocking read from soundcard
if (Capture.portaudio_index < 0)
nSamples = quisk_read_alsa(&Capture, cSamples);
else
nSamples = quisk_read_portaudio(&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);
}
else {
nSamples = 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 samples to a file
if (want_record) {
if (is_recording_samples) {
record_samples(cSamples, nSamples); // Record samples
}
else if (file_name_samples[0]) {
if (record_samples(NULL, -1)) // Open file
is_recording_samples = 1;
}
}
else if (is_recording_samples) {
record_samples(NULL, -2); // Close file
is_recording_samples = 0;
}
#if ! DEBUG_MIC
nSamples = quisk_process_samples(cSamples, nSamples);
#endif
#if DEBUG_IO > 1
QuiskPrintTime(" process samples", 0);
#endif
if (Playback.portaudio_index < 0)
quisk_play_alsa(&Playback, nSamples, cSamples, 1);
else
quisk_play_portaudio(&Playback, nSamples, cSamples, 1);
if (rxMode == 7) {
if (DigitalOutput.portaudio_index < 0)
quisk_play_alsa(&DigitalOutput, nSamples, cSamples, 0);
else
quisk_play_portaudio(&DigitalOutput, nSamples, cSamples, 0);
}
// Perhaps record the speaker audio to a file
if (want_record) {
if (is_recording_audio) {
record_audio(cSamples, nSamples); // Record samples
}
else if (file_name_audio[0]) {
if (record_audio(NULL, -1)) // Open file
is_recording_audio = 1;
}
}
else if (is_recording_audio) {
record_audio(NULL, -2); // Close file
is_recording_audio = 0;
}
#if DEBUG_IO > 1
QuiskPrintTime(" play samples", 0);
#endif
// Read and process the microphone
mic_count = 0;
mic_sample_rate = quisk_sound_state.mic_sample_rate;
if (MicCapture.handle) {
if (MicCapture.portaudio_index < 0)
mic_count = quisk_read_alsa(&MicCapture, cSamples);
else
mic_count = quisk_read_portaudio(&MicCapture, cSamples);
}
if (quisk_record_state == PLAYBACK) { // Discard previous samples and replace with saved sound
quisk_tmp_microphone(cSamples, mic_count);
}
if (rxMode == 7) { // Discard previous samples and use digital samples
if (DigitalInput.handle) {
mic_sample_rate = DigitalInput.sample_rate;
if (DigitalInput.portaudio_index < 0)
mic_count = quisk_read_alsa(&DigitalInput, cSamples);
else
mic_count = quisk_read_portaudio(&DigitalInput, cSamples);
}
else {
mic_count = 0;
}
}
if (mic_count > 0) {
#if DEBUG_IO > 1
QuiskPrintTime(" mic-read", 0);
#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
if ( ! quisk_is_key_down())
for (i = 0; i < mic_count; i++)
cSamples[i] = 0;
for (i = 0; i < mic_count; i++)
cSamples[i] *= (double)CLIP32 / CLIP16; // convert 16-bit samples to 32 bits
quisk_process_samples(cSamples, mic_count);
#endif
#if DEBUG_IO > 1
QuiskPrintTime(" mic-proc", 0);
#endif
}
// Mic playback without a mic is needed for CW
if (MicPlayback.handle) { // Mic playback: send mic I/Q samples to a sound card
if (rxMode == 0 || rxMode == 1) { // 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;
}
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 { // Transmit SSB
if ( ! quisk_is_key_down()) {
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
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
if (MicPlayback.portaudio_index < 0)
quisk_play_alsa(&MicPlayback, mic_count, cSamples, 0);
else
quisk_play_portaudio(&MicPlayback, mic_count, cSamples, 0);
#if DEBUG_MIC == 2
quisk_process_samples(cSamples, mic_count);
#endif
}
#if DEBUG_IO > 1
QuiskPrintTime(" finished", 0);
#endif
// Return negative number for error
return retval;
}
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;
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};
key_state = quisk_is_key_down(); //reading this once is important for predicable bevavior on cork/flush
#if DEBUG_MIC == 1
complex double tmpSamples[SAMP_BUFFER_SIZE];
#endif
quisk_sound_state.interupts++;
#if DEBUG_IO > 1
QuiskPrintTime("Start read_sound", 0);
#endif
if (pt_sample_read) { // read samples from SDR-IQ or UDP
nSamples = (*pt_sample_read)(cSamples);
}
else if (Capture.handle) { // blocking read from soundcard
if (Capture.driver == DEV_DRIVER_PULSEAUDIO && quisk_sound_state.IQ_server[0]) {
if (key_state == 1 && !(Capture.cork_status))
quisk_cork_pulseaudio(&Capture, 1);
else if (key_state == 0 && Capture.cork_status) {
quisk_flush_pulseaudio(&Capture);
quisk_cork_pulseaudio(&Capture, 0);
}
}
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);
}
else {
nSamples = 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 samples to a file
if (want_record) {
if (is_recording_samples) {
record_samples(cSamples, nSamples); // Record samples
}
else if (file_name_samples[0]) {
if (record_samples(NULL, -1)) // Open file
is_recording_samples = 1;
}
}
else if (is_recording_samples) {
record_samples(NULL, -2); // Close file
is_recording_samples = 0;
}
// Perhaps write samples to a loopback device for use by another program
if (RawSamplePlayback.handle)
play_sound_interface(&RawSamplePlayback, nSamples, cSamples, 0, 1.0);
#if ! DEBUG_MIC
nSamples = quisk_process_samples(cSamples, nSamples);
#endif
#if DEBUG_IO > 1
QuiskPrintTime(" process samples", 0);
#endif
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
play_sound_interface(&Playback, nSamples, cSamples, 1, quisk_audioVolume);
// Play digital if required
if (rxMode == 7 || rxMode == 8 || rxMode == 9)
play_sound_interface(&DigitalOutput, nSamples, cSamples, 0, digital_output_level);
// Perhaps record the speaker audio to a file
if (want_record) {
if (is_recording_audio) {
record_audio(cSamples, nSamples); // Record samples
}
else if (file_name_audio[0]) {
if (record_audio(NULL, -1)) // Open file
is_recording_audio = 1;
}
}
else if (is_recording_audio) {
record_audio(NULL, -2); // Close file
is_recording_audio = 0;
}
#if DEBUG_IO > 1
QuiskPrintTime(" play samples", 0);
#endif
// Read and process the microphone
mic_count = 0;
mic_sample_rate = quisk_sound_state.mic_sample_rate;
if (MicCapture.handle)
mic_count = read_sound_interface(&MicCapture, cSamples);
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 (rxMode == 7 || rxMode == 8 || rxMode == 9) { // Discard previous samples and use digital samples
if (DigitalInput.handle) {
if (DigitalInput.driver == DEV_DRIVER_PULSEAUDIO && quisk_sound_state.IQ_server[0]) {
if (key_state == 0 && !(DigitalInput.cork_status)) {//key is up, cork stream
quisk_cork_pulseaudio(&DigitalInput, 1);
}
else if (key_state == 1 && DigitalInput.cork_status) { //key is down, uncork and flush
quisk_flush_pulseaudio(&DigitalInput);
if (MicPlayback.handle)
quisk_flush_pulseaudio(&MicPlayback);
quisk_cork_pulseaudio(&DigitalInput, 0);
}
}
mic_sample_rate = DigitalInput.sample_rate;
mic_count = read_sound_interface(&DigitalInput, cSamples);
}
else {
mic_count = 0;
}
}
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
}
// Mic playback without a mic is needed for CW
if (MicPlayback.handle) { // Mic playback: send mic I/Q samples to a sound card
if (rxMode == 0 || rxMode == 1) { // Transmit CW
is_cw = 1;
if (MicCapture.driver == DEV_DRIVER_PULSEAUDIO && MicCapture.cork_status) {
//uncork and flush streams. Don't cork on key-up while in CW mode.
quisk_flush_pulseaudio(&MicCapture);
quisk_cork_pulseaudio(&MicCapture, 0);
quisk_flush_pulseaudio(&MicPlayback);
}
}
else {
is_cw = 0;
cwCount = 0;
cwEnvelope = 0.0;
//Not in CW mode, cork/uncork capture streams as PTT is used.
if (MicCapture.driver == DEV_DRIVER_PULSEAUDIO && MicCapture.handle && quisk_sound_state.IQ_server[0]) {
if (key_state == 0 && !(MicCapture.cork_status)) {//key is up, cork stream
quisk_cork_pulseaudio(&MicCapture, 1);
}
else if (key_state == 1 && DigitalInput.cork_status) { //key is down, uncork and flush
quisk_flush_pulseaudio(&MicCapture);
quisk_flush_pulseaudio(&MicPlayback);
quisk_cork_pulseaudio(&MicCapture, 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()) { // Transmit SSB
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
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
play_sound_interface(&MicPlayback, mic_count, cSamples, 0, 1.0);
#if DEBUG_MIC == 2
quisk_process_samples(cSamples, mic_count);
#endif
}
prev_key_state = key_state;
#if DEBUG_IO > 1
QuiskPrintTime(" finished", 0);
#endif
// Return negative number for error
return retval;
}
void handle_recordstate(MENU *state) {
#ifdef TESTMODE
printf("Entered record state\n");
#endif
TEXT output;
output.row = 0;
output.column = 0;
output.size = 1;
colorcopy(output.RGB, red);
nextState = STATE_NULL;
int stop = 0;
while (stop == 0) {
keysWaitOnPress();
if ((keysGetLeft() == 1) || (keysGetRight() == 1)) {
printf("buttonpress\n");
state->activeButton += 1;
if (state->activeButton == state->buttonCount) {
state->activeButton = 0;
}
updateButtonSelection(state);
refreshButtons(state);
}
if (keysGetEnter() == 1) {
printf("ENTER\n");
switch (state->activeButton) {
case 0:
//get global settings
//--start of temporary data--//
audio_set.duration = 0.250;
audio_set.numChannels = 1;
audio_set.sampleRate = 44100;
audio_set.numSamples = audio_set.duration
* audio_set.sampleRate;
//--end of temporary data----//
TEXT output;
output.row = 0;
output.column = 0;
output.size = 2;
colorcopy(output.RGB, red);
//-----record audio-----
sprintf(output.text, "RECORDING 250ms of NOISE\n");
draw_text(output);
output.row++;
if (record_audio(noiseFile, &audio_set) < 0) {
//error doorgeven
nextState = STATE_NULL;
break;
}
printf("record done");
sprintf(output.text, "RECORDING\n");
draw_text(output);
output.row++;
audio_set.duration = 0.450;
if (record_audio(fileName, &audio_set) < 0) {
//error doorgeven
nextState = STATE_NULL;
break;
}
printf("record done");
nextState = STATE_CALCULATE;
stop = 1;
break;
default:
nextState = STATE_MAIN;
stop = 1;
break;
}
}
}
}