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);
// 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, "IQ Input", QUISK_SC_SIZE);
Capture.stream_description[QUISK_SC_SIZE-1] = '\0';
strncpy(Playback.stream_description, "Radio Audio 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, "IQ Output", QUISK_SC_SIZE);
MicPlayback.stream_description[QUISK_SC_SIZE-1] = '\0';
strncpy(DigitalInput.stream_description, "Digital Input", QUISK_SC_SIZE);
strncpy(DigitalOutput.stream_description, "Digital Output", QUISK_SC_SIZE);
strncpy(RawSamplePlayback.stream_description, "Raw Sample 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;
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);
//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;
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
// set capture and playback for raw samples
RawSamplePlayback.read_frames = 0;
RawSamplePlayback.latency_frames = RawSamplePlayback.sample_rate * 500 / 1000; // 500 milliseconds
#if DEBUG_IO
printf("Sample buffer size %d, latency msec %d\n", SAMP_BUFFER_SIZE, quisk_sound_state.latency_millisecs);
#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;
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);
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 sound for DGTL
strncpy(DigitalInput.name, QuiskGetConfigString ("digital_input_name", ""), QUISK_SC_SIZE);
DigitalInput.sample_rate = 48000;
DigitalInput.channel_I = 0;
DigitalInput.channel_Q = 1;
// Playback device for digital sound for DGTL
strncpy(DigitalOutput.name, QuiskGetConfigString ("digital_output_name", ""), QUISK_SC_SIZE);
DigitalOutput.sample_rate = quisk_sound_state.playback_rate; // Radio sound play rate
DigitalOutput.channel_I = 0;
DigitalOutput.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);
#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
Capture.name[0] = 0; // zero the capture soundcard name
}
else if (quisk_using_udp) { // samples from UDP at multiple of 48 kHz
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 mode DGTL
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
#if DEBUG_IO
printf("Sample buffer size %d, latency msec %d\n", SAMP_BUFFER_SIZE, quisk_sound_state.latency_millisecs);
#endif
}