struct FDMDV_STATS g_stats;
int g_nSoundCards = 2;
+
int g_soundCard1InDeviceNum = 0;
int g_soundCard1OutDeviceNum = 0;
+int g_soundCard1SampleRate = 48000;
+
int g_soundCard2InDeviceNum = 1;
int g_soundCard2OutDeviceNum = 1;
+int g_soundCard2SampleRate = 44100;
-int cb_cnt;
+int cb_cnt, cb1, cb2;
int mute_mic = 0;
int read_file = 0;
FILE *g_file;
fifo_destroy(m_rxUserdata->rxoutfifo);
}
-int MainFrame::initPortAudioDevice(PortAudioWrap *pa, int inDevice, int outDevice, int soundCard)
+int MainFrame::initPortAudioDevice(PortAudioWrap *pa, int inDevice, int outDevice, int soundCard, int sampleRate)
{
char s[256];
equal PA_PFB, for example when I set PA_FPB to 960 I got
framesPerBuffer = 1024.
*/
+ printf("Sound Card %d Sample rate %d\n", soundCard, sampleRate);
pa->setFramesPerBuffer(PA_FPB);
- pa->setSampleRate(SAMPLE_RATE);
+ pa->setSampleRate(sampleRate);
pa->setStreamFlags(0);
return 0;
//-------------------------------------------------------------------------
void MainFrame::startRxStream()
{
+ int src_error;
+
if(!m_RxRunning)
{
cb_cnt = 0;
m_rxDevOut = m_rxPa->getDefaultOutputDevice();
}
- if (initPortAudioDevice(m_rxPa, m_rxDevIn, m_rxDevOut, 1) != 0) {
+ if (initPortAudioDevice(m_rxPa, m_rxDevIn, m_rxDevOut, 1, g_soundCard1SampleRate) != 0) {
delete m_rxPa;
m_RxRunning = false;
return;
m_txDevIn = g_soundCard2InDeviceNum;
m_txDevOut = g_soundCard2OutDeviceNum;
- if (initPortAudioDevice(m_txPa, m_txDevIn, m_txDevOut, 2) != 0) {
+ if (initPortAudioDevice(m_txPa, m_txDevIn, m_txDevOut, 2, g_soundCard2SampleRate) != 0) {
delete m_rxPa;
delete m_txPa;
m_RxRunning = false;
for(int i = 0; i < MEM8; i++)
{
- m_rxUserdata->in8k1[i] = (float)0.0;
m_rxUserdata->in8k2[i] = (float)0.0;
}
for(int i = 0; i < FDMDV_OS_TAPS; i++)
{
m_rxUserdata->in48k1[i] = (float)0.0;
- m_rxUserdata->in48k2[i] = (float)0.0;
}
+ m_rxUserdata->insrc2 = src_new(SRC_SINC_FASTEST, 1, &src_error);
+ assert(m_rxUserdata->insrc2 != NULL);
+ m_rxUserdata->outsrc2 = src_new(SRC_SINC_FASTEST, 1, &src_error);
+ assert(m_rxUserdata->outsrc2 != NULL);
+
// create FIFOs used to interface between different buffer sizes
m_rxUserdata->infifo1 = fifo_create(2*N48);
- m_rxUserdata->outfifo2 = fifo_create(4*N48);
+ m_rxUserdata->outfifo1 = fifo_create(4*N48);
/*
During soundcard 1 callback, outfifo1 is read in PA_FPB = 1024 sample blocks.
+ so lets make it 1920*2 in size
*/
- m_rxUserdata->outfifo1 = fifo_create(4*N48);
+ m_rxUserdata->outfifo2 = fifo_create(4*N48);
/*
infifo2 holds buffers from the microphone. These get read
// 48 to 8 kHz sample rate conversion filter states
- float *in8k1 = cbData->in8k1;
- float *in48k1 = cbData->in48k1;
float *in8k2 = cbData->in8k2;
- float *in48k2 = cbData->in48k2;
+ float *in48k1 = cbData->in48k1;
// temp buffers re-used by tx and rx processing
short in8k_short[N8];
+ float in8k[N8];
float out8k[2*N8];
- short out8k_short[N8];
+ short out8k_short[2*N8];
float out48k[2*N48];
short out48k_short[2*N48];
short in48k_short[2*N48];
+ float in48k[2*N48];
short indata[MAX_FPB];
short outdata[MAX_FPB];
+ SRC_DATA src_data;
int ret;
unsigned int i;
(void) timeInfo;
(void) statusFlags;
+ //printf("%d cb1 .. %d", cb1++, cb1-cb2);
assert(inputBuffer != NULL);
assert(outputBuffer != NULL);
/*
// headset aid in tuning (ie we hear SSB radio output)
if (g_State == 0)
- memcpy(&in8k1[MEM8], out8k, sizeof(float)*N8);
+ memcpy(in8k, out8k, sizeof(float)*N8);
else {
memset(in8k_short, 0, sizeof(short)*N8);
fifo_read(cbData->rxoutfifo, in8k_short, N8);
- short_to_float(&in8k1[MEM8], in8k_short, N8);
+ src_short_to_float_array(in8k_short, in8k, N8);
+ //short_to_float(&in8k1[MEM8], in8k_short, N8);
}
- fdmdv_8_to_48(out48k, &in8k1[MEM8], N8);
- float_to_short(out48k_short, out48k, N48);
- if (g_nSoundCards == 1)
- fifo_write(cbData->outfifo1, out48k_short, N48);
+ //fdmdv_8_to_48(out48k, &in8k1[MEM8], N8);
+ //float_to_short(out48k_short, out48k, N48);
+ src_data.data_in = in8k;
+ src_data.data_out = out48k;
+ src_data.input_frames = N8;
+ src_data.output_frames = N48;
+ src_data.end_of_input = 0;
+
+ if (g_nSoundCards == 1) {
+ src_data.src_ratio = (float)g_soundCard1SampleRate/FS;
+ src_process(cbData->outsrc2, &src_data);
+ //printf("src_data.output_frames_gen: %d\n", src_data.output_frames_gen);
+ src_float_to_short_array(out48k, out48k_short, src_data.output_frames_gen);
+ fifo_write(cbData->outfifo1, out48k_short, src_data.output_frames_gen);
+ }
else {
- //printf("sc1: fifo write n before = %d", fifo_n(cbData->outfifo2));
- fifo_write(cbData->outfifo2, out48k_short, N48);
- //printf(" after = %d\n", fifo_n(cbData->outfifo2));
+ src_data.src_ratio = (float)g_soundCard2SampleRate/FS;
+ src_process(cbData->outsrc2, &src_data);
+ src_float_to_short_array(out48k, out48k_short, src_data.output_frames_gen);
+ fifo_write(cbData->outfifo2, out48k_short, src_data.output_frames_gen);
+ //printf("after outfifo2 write %d fifo_n: %d\n", src_data.output_frames_gen, fifo_n(cbData->outfifo2));
}
+
+ assert(src_data.output_frames_gen <= N48);
}
//
short tx_speech_in[2*N8];
short tx_mod_out[2*N8];
+ // number of microphone samples we need at g_soundcards2SampleRate
+
+ int nsam = g_soundCard2SampleRate * (float)codec2_samples_per_frame(g_pCodec2)/FS;
+ assert(nsam <= 2*N48);
+
// infifo2 is written to by another sound card so it may
// over or underflow, but we don't realy care. It will
// just result in a short interruption in audio being fed
// to codec2_enc, possibly making a click every now and
// again in the decoded audio at the other end.
+
// zero speech input just in case infifo2 underflows
- memset(in48k_short, 0, sizeof(short)*2*N48);
+ memset(in48k_short, 0, sizeof(short)*nsam);
- // Codec 2 @ 1400 bit/s requires 40ms of input speech
+ fifo_read(cbData->infifo2, in48k_short, nsam);
- fifo_read(cbData->infifo2, in48k_short, 2*N48);
-
if (mute_mic)
- memset(in48k_short, 0, sizeof(short)*2*N48);
- short_to_float(&in48k2[FDMDV_OS_TAPS], in48k_short, 2*N48);
- fdmdv_48_to_8(out8k, &in48k2[FDMDV_OS_TAPS], 2*N8);
- float_to_short(tx_speech_in, out8k, 2*N8);
- /*
+ memset(in48k_short, 0, sizeof(short)*nsam);
+
+ src_short_to_float_array(in48k_short, in48k, nsam);
+
+ src_data.data_in = in48k;
+ src_data.data_out = out8k;
+ src_data.input_frames = nsam;
+ src_data.output_frames = codec2_samples_per_frame(g_pCodec2);
+ src_data.end_of_input = 0;
+ src_data.src_ratio = (float)FS/g_soundCard2SampleRate;
+
+ src_process(cbData->insrc2, &src_data);
+ assert(src_data.output_frames_gen <= codec2_samples_per_frame(g_pCodec2));
+
+ src_float_to_short_array(out8k, tx_speech_in, src_data.output_frames_gen);
+
+ if (write_file) {
+ fwrite(tx_speech_in, sizeof(short), 2*N8, g_write_file);
+ }
+
if (read_file) {
int n = fread( tx_speech_in, sizeof(short), 2*N8, g_file);
if (n != 2*N8) {
rewind(g_file);
}
}
- */
+
assert(codec2_samples_per_frame(g_pCodec2) == (2*N8));
per_frame_tx_processing(tx_mod_out, tx_speech_in, g_pCodec2);
short_to_float(&in8k2[MEM8], tx_mod_out, 2*N8);
fdmdv_8_to_48(out48k, &in8k2[MEM8], 2*N8);
- float_to_short(out48k_short, out48k, 2*N48);
+ float_to_short(out48k_short, out48k, 2*N48);
fifo_write(cbData->outfifo1, out48k_short, 2*N48);
}
}
wptr[1] = 0;
}
}
+ //printf("end cb1\n");
return paContinue;
}
// add decoded speech to end of output buffer
assert(codec2_samples_per_frame(c2) == (2*N8));
- {
- short input_buf[2*N8];
- if (read_file) {
- int n = fread(input_buf , sizeof(short), 2*N8, g_file);
- if (n != 2*N8) {
- //printf("rewind\n");
- rewind(g_file);
- }
- }
- codec2_encode(c2, packed_bits, input_buf);
- }
codec2_decode(c2, output_buf, packed_bits);
- //#define SINE
-#ifdef SINE
- {
- float two_pi = 8.0*atan(1);
- for(i=0; i<codec2_samples_per_frame(c2); i++)
- output_buf[i] = 1000.0 * cos(two_pi*i/16.0);
- }
-#endif
fifo_write(output_fifo, output_buf, codec2_samples_per_frame(c2));
}
int sync_bit;
int i, bit, byte;
- //codec2_encode(c2, packed_bits, input_buf);
+ codec2_encode(c2, packed_bits, input_buf);
/* unpack bits, MSB first */
short indata[MAX_FPB];
short outdata[MAX_FPB];
+ //printf("%d cb2 ... %d", cb2++,cb1-cb2);
+
// assemble a mono buffer (just use left channel) and write to FIFO
assert(framesPerBuffer < MAX_FPB);
indata[i] = *rptr;
}
-//#define SC2_LOOPBACK
+ //#define SC2_LOOPBACK
#ifdef SC2_LOOPBACK
for(i = 0; i < framesPerBuffer; i++, wptr += 2)
{
if (fifo_read(cbData->outfifo2, outdata, framesPerBuffer) == 0)
{
- if (write_file) {
- fwrite(outdata, sizeof(short), framesPerBuffer, g_write_file);
- }
// write signal to both channels */
for(i = 0; i < framesPerBuffer; i++, wptr += 2)
{
}
}
#endif
+
+ //printf("end cb2\n");
return paContinue;
}
projects / freetel-svn-tracking.git / commitdiff