int g_nRxIn = FDMDV_NOM_SAMPLES_PER_FRAME;
float g_Ts = 0.0;
-short g_RxInBuf[2 * FDMDV_NOM_SAMPLES_PER_FRAME];
-short *g_pRxOutBuf;
-int g_nInputBuf = 0;
-
-short g_TxInBuf[2 * FDMDV_NOM_SAMPLES_PER_FRAME];
-short *g_pTxOutBuf;
-int g_nOutputBuf = 0;
-
-
int g_CodecBits[2 * FDMDV_BITS_PER_FRAME];
int g_State = 0;
m_RxRunning = true;
m_rxPa = new PortAudioWrap();
-#ifdef _DUMMY_DATA
- for(int i = 0; i < FDMDV_NSPEC; i++)
- {
-// m_rxPa->m_av_mag[i] = sin(((double)i / M_PI)) * 100.0;
- g_avmag[i] = (float)sin(((double)i / M_PI)) * 100.0;
- }
-#else
- for(int i = 0; i < FDMDV_NSPEC; i++)
- {
-// m_rxPa->m_av_mag[i] = -40.0;
- g_avmag[i] = (float)-40.0;
- }
-#endif // _DUMMY_DATA
-
- g_pRxOutBuf = (short*)malloc(2 * sizeof(short) * codec2_samples_per_frame(g_pCodec2));
-
m_rxDevIn = m_rxPa->getDefaultInputDevice(); // default input device
if(m_rxDevIn == paNoDevice)
{
{
wxMessageBox(wxT("Rx Error: No default output device."), wxT("Error"), wxOK);
delete m_rxPa;
- free(g_pRxOutBuf);
m_RxRunning = false;
return;
}
m_rxUserdata->infifo = fifo_create(2*N48);
m_rxUserdata->outfifo = fifo_create(2*N48);
+ m_rxUserdata->rxinfifo = fifo_create(2 * FDMDV_NOM_SAMPLES_PER_FRAME);
+ m_rxUserdata->rxoutfifo = fifo_create(2 * codec2_samples_per_frame(g_pCodec2));
m_rxPa->setUserData(m_rxUserdata);
m_rxErr = m_rxPa->setCallback(rxCallback);
delete m_rxPa;
fifo_destroy(m_rxUserdata->infifo);
fifo_destroy(m_rxUserdata->outfifo);
- free(g_pRxOutBuf);
+ fifo_destroy(m_rxUserdata->rxinfifo);
+ fifo_destroy(m_rxUserdata->rxoutfifo);
return;
}
m_rxErr = m_rxPa->streamStart();
delete m_rxPa;
fifo_destroy(m_rxUserdata->infifo);
fifo_destroy(m_rxUserdata->outfifo);
- free(g_pRxOutBuf);
+ fifo_destroy(m_rxUserdata->rxinfifo);
+ fifo_destroy(m_rxUserdata->rxoutfifo);
return;
}
printf("end startRxStream\n");
delete m_rxPa;
fdmdv_destroy(g_pFDMDV);
codec2_destroy(g_pCodec2);
- free(g_pRxOutBuf);
fifo_destroy(m_rxUserdata->infifo);
fifo_destroy(m_rxUserdata->outfifo);
+ fifo_destroy(m_rxUserdata->rxinfifo);
+ fifo_destroy(m_rxUserdata->rxoutfifo);
delete m_rxUserdata;
}
/*
return;
}
- g_pTxOutBuf = (short*)malloc(2*sizeof(short)*codec2_samples_per_frame(g_pCodec2));
-
m_txErr = m_txPa->setInputDevice(m_txDevIn);
m_txErr = m_txPa->setInputChannelCount(2); // stereo input
m_txErr = m_txPa->setInputSampleFormat(PA_SAMPLE_TYPE);
for(i = 0; i < FDMDV_NSPEC; i++)
{
-// m_rxPa->m_av_mag[i] = (1.0 - BETA) * m_rxPa->m_av_mag[i] + BETA * mag_dB[i];
g_avmag[i] = (1.0 - BETA) * g_avmag[i] + BETA * mag_dB[i];
}
}
+inline void short_to_float(float out_float[], short in_short[], int n) {
+ int i;
+ for(i=0; i<n; i++)
+ out_float[i] = (float)in_short[i];
+}
+
+inline void float_to_short(short out_short[], float in_float[], int n) {
+ int i;
+ for(i=0; i<n; i++)
+ out_short[i] = (short)in_float[i];
+}
+
//-------------------------------------------------------------------------
// rxCallback()
//-------------------------------------------------------------------------
short *wptr = (short*)outputBuffer;
float *in8k = cbData->in8k;
float *in48k = cbData->in48k;
+ short in8k_short[N8];
float out8k[N8];
+ short out8k_short[N8];
float out48k[N48];
short out48k_short[N48];
short in48k_short[N48];
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.
+ we do a little FIFO buffering to interface the different input
+ and output number of sample requirements.
+
+ We also use FIFOs at the input of the demod and output of the
+ decoder to interface between different buffer sizes.
*/
// assemble a mono buffer (just use left channel) and write to FIFO
ret = fifo_write(cbData->infifo, indata, framesPerBuffer);
assert(ret != -1);
- // while we have enough samples available ...
+ // while we have enough input 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
+ // note: modifying fdmdv_48_to_8() to use shorts for sample I/O
+ // would remove all these float to short conversions
+
+ short_to_float(&in48k[FDMDV_OS_TAPS], in48k_short, N48);
fdmdv_48_to_8(out8k, &in48k[FDMDV_OS_TAPS], N8);
- for(i = 0; i < FDMDV_OS_TAPS; i++)
- {
- in48k[i] = in48k[i + N48];
- }
+ float_to_short(out8k_short, out8k, N8);
+ fifo_write(cbData->rxinfifo, out8k_short, N8);
- assert((g_nInputBuf + N8) <= 2 * FDMDV_NOM_SAMPLES_PER_FRAME);
- // 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_nOutputBuf, g_CodecBits, g_RxInBuf, &g_nInputBuf, &g_nRxIn, &g_State, g_pCodec2);
+ per_frame_rx_processing(cbData->rxoutfifo, g_CodecBits, cbData->rxinfifo, &g_nRxIn, &g_State, g_pCodec2);
- // prepare output buffer for D/A (speaker)
- if (g_nOutputBuf >= N8)
- {
- // if demod out of sync copy input audio from A/D to aid in tuning
- if(g_State == 0)
- {
- for(i = 0; i < N8; i++)
- {
- in8k[MEM8 + i] = out8k[i]; // echo 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);
+ // if demod out of sync copy input audio from A/D to aid in tuning
- // shift speech samples in output buffer
- for(i = 0; i < (unsigned int)g_nOutputBuf; i++)
- {
- g_pRxOutBuf[i] = g_pRxOutBuf[i + N8];
- }
+ if (g_State == 0)
+ memcpy(&in8k[MEM8], out8k, sizeof(float)*N8);
+ else {
+ fifo_read(cbData->rxoutfifo, in8k_short, N8);
+ short_to_float(&in8k[MEM8], in8k_short, N8);
+ }
- // test: echo input to output, make this loopback option
- /*
- 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];
- }
- assert(outputBuffer != NULL);
-
- // write signal to fifo
- for(i = 0; i < N48; i++)
- {
- out48k_short[i] = (short)out48k[i];
- }
+ float_to_short(out48k_short, out48k, N48);
fifo_write(cbData->outfifo, out48k_short, N48);
}
// OK now set up output samples
+
if (fifo_read(cbData->outfifo, outdata, framesPerBuffer) == 0)
{
// write signal to both channels */
}
else
{
- //printf("no data\n");
- // zero output if no data available */
+ // zero output if no data available
for(i = 0; i < framesPerBuffer; i++, wptr += 2)
{
wptr[0] = 0;
//----------------------------------------------------------------
// per_frame_rx_processing()
//----------------------------------------------------------------
- void MainFrame::per_frame_rx_processing(
- short output_buf[], // output buf of decoded speech samples
- int *n_output_buf, // how many samples currently in output_buf[]
- int codec_bits[], // current frame of bits for decoder
- short input_buf[], // input buf of modem samples input to demod
- int *n_input_buf, // how many samples currently in input_buf[]
+void MainFrame::per_frame_rx_processing(
+ FIFO *output_fifo, // decoded speech samples
+ int codec_bits[], // current frame of bits for decoder
+ FIFO *input_fifo, // modem samples input to demod
int *nin, // amount of samples demod needs for next call
int *state, // used to collect codec_bits[] halves
CODEC2 *c2 // Codec 2 states
)
{
int sync_bit;
+ short input_buf[FDMDV_MAX_SAMPLES_PER_FRAME];
+ short output_buf[N8*2];
float rx_fdm[FDMDV_MAX_SAMPLES_PER_FRAME];
int rx_bits[FDMDV_BITS_PER_FRAME];
unsigned char packed_bits[BYTES_PER_CODEC_FRAME];
int bit;
int byte;
int next_state;
-
- assert(*n_input_buf <= (2 * FDMDV_NOM_SAMPLES_PER_FRAME));
-
+
//
// This while loop will run the demod 0, 1 (nominal) or 2 times:
//
// ppm), case 0 or 2 occured about once every 30 seconds. This is
// no problem for the decoded audio.
//
- while(*n_input_buf >= *nin)
+
+ while (fifo_read(input_fifo, input_buf, *nin) == 0)
{
// demod per frame processing
for(i = 0; i < *nin; i++)
}
nin_prev = *nin;
fdmdv_demod(g_pFDMDV, rx_bits, &sync_bit, rx_fdm, nin);
- *n_input_buf -= nin_prev;
- assert(*n_input_buf >= 0);
-
- // shift input buffer
- for(i = 0; i < *n_input_buf; i++)
- {
- input_buf[i] = input_buf[i + nin_prev];
- }
// compute rx spectrum & get demod stats, and update GUI plot data
fdmdv_get_rx_spectrum(g_pFDMDV, rx_spec, rx_fdm, nin_prev);
{
case 0:
// mute output audio when out of sync
+#ifdef PREV
if(*n_output_buf < 2 * codec2_samples_per_frame(c2) - N8)
{
for(i = 0; i < N8; i++)
*n_output_buf += N8;
}
assert(*n_output_buf <= (2 * codec2_samples_per_frame(c2)));
+#endif
+ 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))
{
next_state = 1;
}
}
assert(byte == BYTES_PER_CODEC_FRAME);
+
// add decoded speech to end of output buffer
- if(*n_output_buf <= codec2_samples_per_frame(c2))
- {
- codec2_decode(c2, &output_buf[*n_output_buf], packed_bits);
- *n_output_buf += codec2_samples_per_frame(c2);
- }
- assert(*n_output_buf <= (2 * codec2_samples_per_frame(c2)));
+
+ assert(codec2_samples_per_frame(c2) <= (2*N8));
+ codec2_decode(c2, output_buf, packed_bits);
+ fifo_write(output_fifo, output_buf, codec2_samples_per_frame(c2));
+
}
break;
}
projects / freetel-svn-tracking.git / commitdiff