//==========================================================================
#ifndef __FDMDV2_DEFINES__
#define __FDMDV2_DEFINES__
+
#include "wx/wx.h"
-/* FFT points */
-#define FDMDV_NSPEC 512
+#include "fdmdv.h"
-#define MIN_DB -40.0
-#define MAX_DB 0.0
-#define BETA 0.1 // constant for time averageing spectrum data
-#define MIN_HZ 0
-#define MAX_HZ 4000
-#define WATERFALL_SECS_Y 5 // number of seconds respresented by y axis of waterfall
-#define DT 0.02 // time between samples
-#define FS 8000 // FDMDV modem sample rate
+// Spectrogram and Waterfall
-#define FDMDV_BITS_PER_FRAME 28 /* 20ms frames, 1400 bit/s */
-#define FDMDV_NOM_SAMPLES_PER_FRAME 160 /* modulator output samples/frame and nominal demod samples/frame */
- /* at 8000 Hz sample rate */
-#define FDMDV_MAX_SAMPLES_PER_FRAME 200 /* max demod samples/frame, use this to allocate storage */
-#define FDMDV_SCALE 1000 /* suggested scaling for 16 bit shorts */
-#define FDMDV_NSYM 15
+#define FDMDV_NSPEC 512
#define MIN_DB -40.0
#define MAX_DB 0.0
#define DT 0.02 // time between samples
#define FS 8000 // FDMDV modem sample rate
+// Scatter diagram
+
#define SCATTER_MEM (FDMDV_NSYM)*50
#define SCATTER_X_MAX 3.0
#define SCATTER_Y_MAX 3.0
#define H2 (H/2)
#define SP 20
-// sound card
+// sample rate I/O & conversion constants
+#define MAX_FPB 2048 // maximum value of framesPerBuffer
+#define PA_FPB 512 // nominal value of framesPerBuffer
#define SAMPLE_RATE 48000 // 48 kHz sampling rate rec. as we can trust accuracy of sound card
#define N8 FDMDV_NOM_SAMPLES_PER_FRAME // processing buffer size at 8 kHz
#define MEM8 (FDMDV_OS_TAPS/FDMDV_OS)
#define BITS_PER_CODEC_FRAME (2 * FDMDV_BITS_PER_FRAME)
#define BYTES_PER_CODEC_FRAME (BITS_PER_CODEC_FRAME / 8)
-/* 8 to 48 kHz sample rate conversion */
-#define FDMDV_OS 6 /* oversampling rate */
-#define FDMDV_OS_TAPS 48 /* number of OS filter taps */
-
enum
{
ID_ROTATE_LEFT = wxID_HIGHEST + 1,
#endif // _USE_TIMER
startRxStream();
// startTxStream();
- m_togBtnOnOff->SetLabel(wxT("Stop"));
+ if (m_RxRunning)
+ m_togBtnOnOff->SetLabel(wxT("Stop"));
}
else
{
if(m_rxDevIn == paNoDevice)
{
wxMessageBox(wxT("Rx Error: No default input device."), wxT("Error"), wxOK);
- return;
+ delete m_rxPa;
+ m_RxRunning = false;
+ return;
}
m_rxErr = m_rxPa->setInputDevice(m_rxDevIn);
m_rxErr = m_rxPa->setInputChannelCount(2); // stereo input
if (m_rxDevOut == paNoDevice)
{
wxMessageBox(wxT("Rx Error: No default output device."), wxT("Error"), wxOK);
+ delete m_rxPa;
+ m_RxRunning = false;
return;
}
m_rxErr = m_rxPa->setOutputDevice(m_rxDevOut);
m_rxErr = m_rxPa->setOutputLatency(m_rxPa->getOutputDefaultLowLatency());
m_rxPa->setOutputHostApiStreamInfo(NULL);
- m_rxErr = m_rxPa->setFramesPerBuffer(FRAMES_PER_BUFFER);
+ m_rxErr = m_rxPa->setFramesPerBuffer(PA_FPB);
m_rxErr = m_rxPa->setSampleRate(SAMPLE_RATE);
m_rxErr = m_rxPa->setStreamFlags(0);
m_rxUserdata->in48k[i] = 0.0;
}
+ m_rxUserdata->infifo = fifo_create(2*N48);
+ m_rxUserdata->outfifo = fifo_create(2*N48);
+
m_rxPa->setUserData(m_rxUserdata);
m_rxErr = m_rxPa->setCallback(rxCallback);
m_rxErr = m_rxPa->streamOpen();
if(m_rxErr != paNoError)
{
wxMessageBox(wxT("Rx Stream Open/Setup error."), wxT("Error"), wxOK);
- return;
+ delete m_rxPa;
+ fifo_destroy(m_rxUserdata->infifo);
+ fifo_destroy(m_rxUserdata->outfifo);
+ m_RxRunning = false;
+ return;
}
m_rxErr = m_rxPa->streamStart();
if(m_rxErr != paNoError)
{
wxMessageBox(wxT("Rx Stream Start Error."), wxT("Error"), wxOK);
- return;
+ delete m_rxPa;
+ fifo_destroy(m_rxUserdata->infifo);
+ fifo_destroy(m_rxUserdata->outfifo);
+ m_RxRunning = false;
+ return;
}
}
}
fdmdv_destroy(g_pFDMDV);
codec2_destroy(g_pCodec2);
// delete g_RxInBuf;
+ fifo_destroy(m_rxUserdata->infifo);
+ fifo_destroy(m_rxUserdata->outfifo);
delete m_rxUserdata;
}
/*
float out8k[N8];
float out48k[N48];
short out48k_short[N48];
+ short in48k_short[N48];
+ short indata[MAX_FPB];
+ short outdata[MAX_FPB];
(void) timeInfo;
(void) statusFlags;
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 lead to framesPerBuffer = 1024.
+
+ To perform the 48 to 8 kHz conversion we need an integer
+ multiple of FDMDV_OS samples to support the interpolation and
+ decimation. As we can't guarantee the size of framesPerBuffer
+ we do a little FIFO buffering.
+ */
+
+ /* assemble a mono buffer (just use left channel) and write to FIFO */
+
+ assert(framesPerBuffer < MAX_FPB);
+ for(i=0; i<framesPerBuffer; i++,rptr+=2)
+ indata[i] = *rptr;
+ fifo_write(cbData->infifo, indata, framesPerBuffer);
+
+ /* while we have enough samples available ... */
+ while (fifo_read(cbData->infifo, in48k_short, N48) == 0) {
+
+ /* convert to float */
+
+ for(i=0; i<N48; i++)
+ in48k[FDMDV_OS_TAPS + i] = in48k_short[i];
+
+ // downsample and update filter memory
+ fdmdv_48_to_8(out8k, &in48k[FDMDV_OS_TAPS], N8);
+ for(i = 0; i < FDMDV_OS_TAPS; i++)
+ {
+ in48k[i] = in48k[i + N48];
+ }
#ifdef RX_CB
- // Convert input model samples from 48 to 8 kHz
- // just use left channel
- for(i = 0; i < framesPerBuffer; i++, rptr += 2)
- {
- in48k[i + FDMDV_OS_TAPS] = *rptr;
- }
- // downsample and update filter memory
- fdmdv_48_to_8(out8k, &in48k[FDMDV_OS_TAPS], N8);
- for(i = 0; i < FDMDV_OS_TAPS; i++)
- {
- in48k[i] = in48k[i + framesPerBuffer];
- }
- // run demod, decoder and update GUI info
- unsigned int j = N8;
- //for(i = 0; i < N8; i++)
- for(i = 0; i < j; i++)
- {
- g_RxInBuf[g_nInputBuf + i] = (short)out8k[i];
- }
- g_nInputBuf += FDMDV_NOM_SAMPLES_PER_FRAME;
- per_frame_rx_processing(g_pRxOutBuf, &g_nInputBuf, g_CodecBits, g_RxInBuf, &g_nOutputBuf, &g_nRxIn, &g_State, g_pCodec2);
- cbData->pWFPanel->m_newdata = true;
- cbData->pSPPanel->m_newdata = true;
- // if demod out of sync copy input audio from A/D to aid in tuning
- if (g_nOutputBuf >= N8)
- {
- if(g_State == 0)
+ // run demod, decoder and update GUI info
+ for(i = 0; i < N8; i++)
+ {
+ g_RxInBuf[g_nInputBuf + i] = (short)out8k[i];
+ }
+ g_nInputBuf += FDMDV_NOM_SAMPLES_PER_FRAME;
+ per_frame_rx_processing(g_pRxOutBuf, &g_nInputBuf, g_CodecBits, g_RxInBuf, &g_nOutputBuf, &g_nRxIn, &g_State, g_pCodec2);
+
+ cbData->pWFPanel->m_newdata = true;
+ cbData->pSPPanel->m_newdata = true;
+
+ // if demod out of sync copy input audio from A/D to aid in tuning
+ if (g_nOutputBuf >= N8)
{
- for(i = 0; i < N8; i++)
+ if(g_State == 0)
{
- in8k[MEM8 + i] = out8k[i]; // A/D signal
- }
- }
- else
+ for(i = 0; i < N8; i++)
+ {
+ in8k[MEM8 + i] = out8k[i]; // A/D signal
+ }
+ }
+ else
+ {
+ for(i = 0; i < N8; i++)
+ {
+ in8k[MEM8+i] = g_pRxOutBuf[i]; // decoded spech
+ }
+ }
+ g_nOutputBuf -= N8;
+ }
+ assert(g_nOutputBuf >= 0);
+ // shift speech samples in output buffer
+ for(i = 0; i < (unsigned int)g_nOutputBuf; i++)
{
- for(i = 0; i < N8; i++)
- {
- in8k[MEM8+i] = g_pRxOutBuf[i]; // decoded spech
- }
+ g_pRxOutBuf[i] = g_pRxOutBuf[i + N8];
+ }
+ #endif
+
+ /* test: echo input to output */
+ for(i=0; i<N8; i++)
+ in8k[MEM8+i] = out8k[i];
+
+ // Convert output speech to 48 kHz sample rate
+ // upsample and update filter memory
+ fdmdv_8_to_48(out48k, &in8k[MEM8], N8);
+ for(i = 0; i < MEM8; i++)
+ {
+ in8k[i] = in8k[i + N8];
}
- g_nOutputBuf -= N8;
- }
- assert(g_nOutputBuf >= 0);
- // shift speech samples in output buffer
- for(i = 0; i < (unsigned int)g_nOutputBuf; i++)
- {
- g_pRxOutBuf[i] = g_pRxOutBuf[i + N8];
- }
- // Convert output speech to 48 kHz sample rate
- // upsample and update filter memory
- fdmdv_8_to_48(out48k, &in8k[MEM8], N8);
- for(i = 0; i < MEM8; i++)
- {
- in8k[i] = in8k[i + N8];
+ assert(outputBuffer != NULL);
+
+ // write signal to fifo
+ for(i = 0; i < N48; i++)
+ {
+ out48k_short[i] = (short)out48k[i];
+ }
+
+ fifo_write(cbData->outfifo, out48k_short, N48);
}
- assert(outputBuffer != NULL);
- // write signal to both channels
- for(i = 0; i < N48; i++)
- {
- out48k_short[i] = (short)out48k[i];
+
+ /* OK now set up output samples */
+
+ if (fifo_read(cbData->outfifo, outdata, framesPerBuffer) == 0) {
+
+ /* write signal to both channels */
+
+ for(i=0; i<framesPerBuffer; i++,wptr+=2) {
+ wptr[0] = outdata[i];
+ wptr[1] = outdata[i];
+ }
}
- for(i = 0; i < framesPerBuffer; i++, wptr += 2)
- {
- wptr[0] = out48k_short[i];
- wptr[1] = out48k_short[i];
+ else {
+ //printf("no data\n");
+ /* zero output if no data available */
+ for(i=0; i<framesPerBuffer; i++,wptr+=2) {
+ wptr[0] = 0;
+ wptr[1] = 0;
+ }
}
-#endif
+
return paContinue;
}
projects / freetel-svn-tracking.git / commitdiff