m_auiNbookCtrl->AddPage(m_panelFreqOffset, L"Frequency \u0394", true, wxNullBitmap);
}
+ wxGetApp().m_framesPerBuffer = pConfig->Read(wxT("/Audio/framesPerBuffer"), PA_FPB);
+
g_soundCard1InDeviceNum = pConfig->Read(wxT("/Audio/soundCard1InDeviceNum"), -1);
g_soundCard1OutDeviceNum = pConfig->Read(wxT("/Audio/soundCard1OutDeviceNum"), -1);
g_soundCard1SampleRate = pConfig->Read(wxT("/Audio/soundCard1SampleRate"), -1);
pConfig->Write(wxT("/Audio/SquelchActive"), g_SquelchActive);
pConfig->Write(wxT("/Audio/SquelchLevel"), (int)(g_SquelchLevel*2.0));
+ pConfig->Write(wxT("/Audio/framesPerBuffer"), wxGetApp().m_framesPerBuffer);
+
pConfig->Write(wxT("/Audio/soundCard1InDeviceNum"), g_soundCard1InDeviceNum);
pConfig->Write(wxT("/Audio/soundCard1OutDeviceNum"), g_soundCard1OutDeviceNum);
pConfig->Write(wxT("/Audio/soundCard1SampleRate"), g_soundCard1SampleRate );
void MainFrame::OnTimer(wxTimerEvent &evt)
{
-
m_panelWaterfall->m_newdata = true;
m_panelWaterfall->Refresh();
if (snr_limited < 0.0) snr_limited = 0;
if (snr_limited > 20.0) snr_limited = 20.0;
- m_gaugeSNR->SetValue((int)(snr_limited+0.5));
+ m_gaugeSNR->SetValue((int)(snr_limited));
// sync LED
m_rbSync->SetForegroundColour( wxColour( 255, 0, 0 ) );
m_rbSync->SetValue(false);
}
-
}
#endif
//----------------------------------------------------------------
void MainFrame::OnIdle(wxIdleEvent& event)
{
- printf("OnIdle\n");
- if(m_panelWaterfall->m_newdata)
- {
- m_panelWaterfall->Refresh();
- }
- if(m_panelSpectrum->m_newdata)
- {
- m_panelSpectrum->Refresh();
+ if (m_RxRunning) {
+ //printf("OnIdle\n");
+ txRxProcessing();
}
}
#endif // _USE_TIMER
g_pFDMDV = fdmdv_create();
g_pCodec2 = codec2_create(CODEC2_MODE_1400);
g_State = 0;
+ printf("g_stats.snr: %f\n", g_stats.snr_est);
// init click-tune states
// portaudio struct so can't return any errors. So no need to trap
// any errors in this function.
+ printf("indDevice: %d outDevice: %d\n", inDevice, outDevice);
// init input params
pa->setInputDevice(inDevice);
// init output params
pa->setOutputDevice(outDevice);
- pa->setOutputChannelCount(2); // stereo input
+ pa->setOutputChannelCount(2); // stereo output
pa->setOutputSampleFormat(PA_SAMPLE_TYPE);
pa->setOutputLatency(pa->getOutputDefaultLowLatency());
pa->setOutputHostApiStreamInfo(NULL);
framesPerBuffer = 1024.
*/
- pa->setFramesPerBuffer(PA_FPB);
+ pa->setFramesPerBuffer(wxGetApp().m_framesPerBuffer);
+ printf("framesPerBuffer: %d\n", wxGetApp().m_framesPerBuffer);
pa->setSampleRate(sampleRate);
- pa->setStreamFlags(0);
+ pa->setStreamFlags(paClipOff);
}
//-------------------------------------------------------------------------
// create FIFOs used to interface between different buffer sizes
- m_rxUserdata->infifo1 = fifo_create(4*N48);
-
- m_rxUserdata->outfifo1 = fifo_create(4*N48);
-
- /*
- During soundcard 1 callback, outfifo1 is read in PA_FPB = 1024 sample blocks.
-
- In tx processing:
- + if outfifo1 has less than framesPerBuffer = PA_FPB = 1024 samples
- + then tx process generates a 40ms @ 48kHz = 1920 sample blocks
- + so we could have 1023 + 1920 samples
- + so lets make it 1920*2 in size
- */
-
- m_rxUserdata->outfifo2 = fifo_create(4*N48);
-
- /*
- infifo2 holds buffers from the microphone. These get read
- in 40ms (1920 sample @ 48 kHz) blocks by the sound card 1
- callback. We write to the sound card in 1024 sample
- blocks. So we need at least 1024+1920 samples, lest also allocate
- 1920*2 samples
- */
-
- m_rxUserdata->infifo2 = fifo_create(4*N48);
+ m_rxUserdata->infifo1 = fifo_create(8*N48);
+ m_rxUserdata->outfifo1 = fifo_create(8*N48);
+ m_rxUserdata->outfifo2 = fifo_create(8*N48);
+ m_rxUserdata->infifo2 = fifo_create(8*N48);
m_rxUserdata->rxinfifo = fifo_create(3 * FDMDV_NOM_SAMPLES_PER_FRAME);
m_rxUserdata->rxoutfifo = fifo_create(2 * codec2_samples_per_frame(g_pCodec2));
fifo_write(plotFifo, dec_samples, nSamples);
}
-//-------------------------------------------------------------------------
-// rxCallback()
-//-------------------------------------------------------------------------
-
-/*
- todo:
-
- + add tests to determine if we have real time audio I/O problems,
- for example lost samples, buffer overflow, too slow processing
- + should flag the user so they can tune the system
- + maybe compute time spent doing demod compared to time between calls
-*/
-
-int MainFrame::rxCallback(
- const void *inputBuffer,
- void *outputBuffer,
- unsigned long framesPerBuffer,
- const PaStreamCallbackTimeInfo* timeInfo,
- PaStreamCallbackFlags statusFlags,
- void *userData
- )
+void MainFrame::txRxProcessing()
{
- paCallBackData *cbData = (paCallBackData*)userData;
- short *rptr = (short*)inputBuffer;
- short *wptr = (short*)outputBuffer;
+ paCallBackData *cbData = m_rxUserdata;
- // temp buffers re-used by tx and rx processing
+ // Buffers re-used by tx and rx processing
// signals in in48k/out48k are at a maximum sample rate of 48k, could be 44.1kHz
// depending on sound hardware.
short in48k_short[2*N48];
short out8k_short[2*N8];
short out48k_short[2*N48];
- short indata[MAX_FPB];
- short outdata[MAX_FPB];
int nout;
- unsigned int i;
-
- (void) timeInfo;
- (void) statusFlags;
-
- //printf("%d cb1 .. %d", cb1++, cb1-cb2);
- assert(inputBuffer != NULL);
- assert(outputBuffer != NULL);
-
- /*
- framesPerBuffer is portaudio-speak for number of samples we
- actually get from the record side and need to provide to the
- play side. On Linux (at least) it was found that
- framesPerBuffer may not always be what we ask for in the
- framesPerBuffer field of Pa_OpenStream. For example a request
- for 960 sample buffers may lead to framesPerBuffer = 1024 in
- this call back.
-
- So we use a bunch of FIFOs to interface between different
- buffer sizes required at each processing step, and to
- communicate between the call backs for the two sound cards.
- */
+ //printf("start infifo1: %5d outfifo1: %5d\n", fifo_n(cbData->infifo1), fifo_n(cbData->outfifo1));
//
// RX side processing --------------------------------------------
//
- // assemble a mono buffer (just use left channel) and write to FIFO
-
- assert(framesPerBuffer < MAX_FPB);
-
- for(i = 0; i < framesPerBuffer; i++, rptr += cbData->inputChannels1)
- {
- indata[i] = *rptr;
- }
- fifo_write(cbData->infifo1, indata, framesPerBuffer);
-
// while we have enough input samples available ...
int nsam = g_soundCard1SampleRate * (float)N8/FS;
if (g_nSoundCards == 2 ) {
- // Make sure we have at least framesPerBuffer modulator output
- // samples. This locks the modulator to the sample
- // rate of sound card 1. We want to make sure that modulator
- // samples are uninterrupted by differences in sample rate
- // between this sound card and sound card 2.
+ // Make sure we have at least 2 frames of modulator output
+ // samples. This locks the modulator to the sample
+ // rate of sound card 1. We want to make sure that modulator
+ // samples are uninterrupted by differences in sample rate
+ // between this sound card and sound card 2.
- while((unsigned)fifo_n(cbData->outfifo1) < framesPerBuffer) {
+ while((unsigned)fifo_n(cbData->outfifo1) < 6*N48) {
int nsam = g_soundCard2SampleRate * (float)codec2_samples_per_frame(g_pCodec2)/FS;
assert(nsam <= 2*N48);
}
}
+ //printf(" end infifo1: %5d outfifo1: %5d\n", fifo_n(cbData->infifo1), fifo_n(cbData->outfifo1));
+
+}
+
+//-------------------------------------------------------------------------
+// rxCallback()
+//-------------------------------------------------------------------------
+
+int MainFrame::rxCallback(
+ const void *inputBuffer,
+ void *outputBuffer,
+ unsigned long framesPerBuffer,
+ const PaStreamCallbackTimeInfo* timeInfo,
+ PaStreamCallbackFlags statusFlags,
+ void *userData
+ )
+{
+ paCallBackData *cbData = (paCallBackData*)userData;
+ short *rptr = (short*)inputBuffer;
+ short *wptr = (short*)outputBuffer;
+
+ short indata[MAX_FPB];
+ short outdata[MAX_FPB];
+
+ unsigned int i;
+
+ (void) timeInfo;
+ (void) statusFlags;
+
+ assert(inputBuffer != NULL);
+ assert(outputBuffer != NULL);
+
+ if (statusFlags)
+ printf("cb1 statusFlags: 0x%x\n", (int)statusFlags);
+
+ //
+ // RX side processing --------------------------------------------
+ //
+
+ // assemble a mono buffer (just use left channel if stereo) and write to FIFO
+
+ assert(framesPerBuffer < MAX_FPB);
+
+ for(i = 0; i < framesPerBuffer; i++, rptr += cbData->inputChannels1)
+ {
+ indata[i] = *rptr;
+ }
+ if (fifo_write(cbData->infifo1, indata, framesPerBuffer)) {
+ //printf("infifo1 full\n");
+ }
+
// OK now set up output samples for this callback
if (fifo_read(cbData->outfifo1, outdata, framesPerBuffer) == 0)
}
else
{
+ //printf("outfifo1 empty\n");
// zero output if no data available
for(i = 0; i < framesPerBuffer; i++, wptr += 2)
{
wptr[1] = 0;
}
}
- //printf("end cb1\n");
return paContinue;
}
// compute rx spectrum & get demod stats, and update GUI plot data
fdmdv_get_rx_spectrum(g_pFDMDV, rx_spec, rx_fdm, nin_prev);
fdmdv_get_demod_stats(g_pFDMDV, &g_stats);
-
+
// Average rx spectrum data using a simple IIR low pass filter
for(i = 0; i < FDMDV_NSPEC; i++)
{
case 0:
// mute output audio when out of sync
- for(i = 0; i < N8; i++)
- output_buf[i] = 0;
- fifo_write(output_fifo, output_buf, N8);
+ for(i = 0; i < N8; i++)
+ output_buf[i] = 0;
+ fifo_write(output_fifo, output_buf, N8);
if((g_stats.fest_coarse_fine == 1) && (g_stats.snr_est > 3.0))
{
short indata[MAX_FPB];
short outdata[MAX_FPB];
- //printf("%d cb2 ... %d", cb2++,cb1-cb2);
+ if (statusFlags)
+ printf("cb2 statusFlags: 0x%x\n", (int)statusFlags);
// assemble a mono buffer (just use left channel) and write to FIFO
assert(framesPerBuffer < MAX_FPB);
- //if (statusFlags)
- // printf("statusFlags: 0x%x\n", statusFlags);
-
for(i = 0; i < framesPerBuffer; i++, rptr += cbData->inputChannels2)
{
indata[i] = *rptr;
//#define SC2_LOOPBACK
#ifdef SC2_LOOPBACK
+
for(i = 0; i < framesPerBuffer; i++, wptr += 2)
{
wptr[0] = indata[i];
wptr[1] = indata[i];
}
+
#else
fifo_write(cbData->infifo2, indata, framesPerBuffer);
projects / freetel-svn-tracking.git / commitdiff