QByteArray &VariCodePipeDecoder::Decode(QVector<short> &bits)
{
decodedbytes.resize(bits.size()+1);
decodedbytes.resize(varicode_decode(&varicode_dec_states,decodedbytes.data(),bits.data(),decodedbytes.size(),bits.size()));//varicode_decode does not work correctly if you say only one output char. is this a bug?
return decodedbytes;
}
void 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[] */
)
{
int sync_bit;
COMP rx_fdm[FDMDV_MAX_SAMPLES_PER_FRAME];
int rx_bits[FDMDV_BITS_PER_FRAME];
unsigned char packed_bits[BYTES_PER_CODEC_FRAME];
float rx_spec[FDMDV_NSPEC];
int i, nin_prev, bit, 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:
0: when tx sample clock runs faster than rx, occasionally we
will run out of samples
1: normal, run decoder once, every 2nd frame output a frame of
speech samples to D/A
2: when tx sample clock runs slower than rx, occasionally we will
have enough samples to run demod twice.
With a +/- 10 Hz sample clock difference at FS=8000Hz (+/- 1250
ppm), case 0 or 1 occured about once every 30 seconds. This is
no problem for the decoded audio.
*/
while(*n_input_buf >= g_nin) {
// demod per frame processing
for(i=0; i<g_nin; i++) {
rx_fdm[i].real = (float)input_buf[i]/FDMDV_SCALE;
rx_fdm[i].imag = 0.0;
}
nin_prev = g_nin;
fdmdv_demod(fdmdv, rx_bits, &sync_bit, rx_fdm, &g_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(fdmdv, rx_spec, rx_fdm, nin_prev);
// Average rx spectrum data using a simple IIR low pass filter
for(i = 0; i < FDMDV_NSPEC; i++)
{
g_avmag[i] = BETA * g_avmag[i] + (1.0 - BETA) * rx_spec[i];
}
fdmdv_get_demod_stats(fdmdv, &stats);
jni_update_stats(&stats, g_avmag);
count++;
/*
State machine to:
+ Mute decoded audio when out of sync. The demod is synced
when we are using the fine freq estimate and SNR is above
a thresh.
+ Decode codec bits only if we have a 0,1 sync bit
sequence. Collects two frames of demod bits to decode
one frame of codec bits.
*/
next_state = g_state;
switch (g_state) {
case 0:
/* mute output audio when out of sync */
if (*n_output_buf < 2*codec2_samples_per_frame(codec2) - N8) {
for(i=0; i<N8; i++)
output_buf[*n_output_buf + i] = 0;
*n_output_buf += N8;
}
if (!(*n_output_buf <= (2*codec2_samples_per_frame(codec2)))) {
LOGE("*n_output_buf <= (2*codec2_samples_per_frame(codec2))");
}
if ((stats.fest_coarse_fine == 1))// && (stats.snr_est > 3.0))
next_state = 1;
break;
case 1:
if (sync_bit == 0) {
next_state = 2;
/* first half of frame of codec bits */
memcpy(codec_bits, rx_bits, FDMDV_BITS_PER_FRAME*sizeof(int));
}
else
next_state = 1;
if (stats.fest_coarse_fine == 0)
next_state = 0;
break;
case 2:
next_state = 1;
if (stats.fest_coarse_fine == 0)
next_state = 0;
if (sync_bit == 1) {
/* second half of frame of codec bits */
memcpy(&codec_bits[FDMDV_BITS_PER_FRAME], rx_bits,
FDMDV_BITS_PER_FRAME*sizeof(int));
// extract data bit
int data_flag_index = codec2_get_spare_bit_index(codec2);
assert(data_flag_index != -1); // not supported for all rates
short abit = codec_bits[data_flag_index];
char ascii_out;
int n_ascii = varicode_decode(&g_varicode_dec_states,
&ascii_out, &abit, 1, 1);
assert((n_ascii == 0) || (n_ascii == 1));
if (n_ascii) {
short ashort = ascii_out;
LOGD("%c", ashort);
}
// reconstruct missing bit we steal for data bit and decode
// speech
codec2_rebuild_spare_bit(codec2, codec_bits);
/* pack bits, MSB received first */
bit = 7; byte = 0;
memset(packed_bits, 0, BYTES_PER_CODEC_FRAME);
for(i=0; i<BITS_PER_CODEC_FRAME; i++) {
packed_bits[byte] |= (codec_bits[i] << bit);
bit--;
if (bit < 0) {
bit = 7;
byte++;
}
}
assert(byte == BYTES_PER_CODEC_FRAME);
/* add decoded speech to end of output buffer */
if (*n_output_buf <= codec2_samples_per_frame(codec2)) {
codec2_decode(codec2, &output_buf[*n_output_buf], packed_bits);
*n_output_buf += codec2_samples_per_frame(codec2);
}
assert(*n_output_buf <= (2*codec2_samples_per_frame(codec2)));
}
break;
}
if (!!g_state != !!next_state) {
jni_update_sync(g_state == 0);
}
g_state = next_state;
}
}