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Fix M17 3200bps Voice; Misc M17 Tweaks; #152

This commit is contained in:
lwvmobile 2023-07-07 02:17:50 -04:00
parent 92a2217e80
commit fb93289538
3 changed files with 44 additions and 37 deletions
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9
src/dsd_frame_sync.c
View File

@ -458,15 +458,10 @@ getFrameSync (dsd_opts * opts, dsd_state * state)
//if using an rtl input method, do not look for sync patterns if the rms value is lower than our 'soft squelch' level
if (opts->audio_in_type == 3 && opts->rtl_rms < opts->rtl_squelch_level) //tests show floor level around 40, and signal breaking 100, default is 100 for level
if (opts->audio_in_type == 3 && opts->rtl_rms < opts->rtl_squelch_level)
{
if (opts->frame_nxdn48 == 1 || opts->frame_nxdn96 == 1 || opts->frame_dpmr == 1)
if (opts->frame_nxdn48 == 1 || opts->frame_nxdn96 == 1 || opts->frame_dpmr == 1 || opts->frame_m17)
{
//should we update min/max here? yes or no?
// state->max = ((state->max) + lmax) / 2;
// state->min = ((state->min) + lmin) / 2;
goto SYNC_TEST_END;
}
}

6
src/dsd_main.c
View File

@ -2472,7 +2472,11 @@ main (int argc, char **argv)
// }
int fmt;
int speed = 48000;
int speed = 48000;
//Codec2 only running at 8k output currently, need to match for OSS
if (opts.frame_m17 == 1)
speed = 8000;
//NOTE: Both /dev/audio AND /dev/dsp randomly open multiple input streams in Linux under padsp wrapper
if((strncmp(opts.audio_in_dev, "/dev/audio", 10) == 0))

66
src/m17.c
View File

@ -142,7 +142,7 @@ int M17processLICH(dsd_state * state, dsd_opts * opts, uint8_t lich_bits[96])
uint8_t lich_reserve = (uint8_t)ConvertBitIntoBytes(&lich_decoded[43], 5); //lich_reserved
if (err == 0)
fprintf (stderr, "LC: %d/6 RES: %02X ", lich_counter+1, lich_reserve); //do the same thing we did on fusion and nxdn
fprintf (stderr, "LC: %d/6 ", lich_counter+1);
else fprintf (stderr, "LICH G24 ERR");
//this is what 2.rrc is supposed to be -- according to m17-demod -- code fixed now and reflects this
@ -190,55 +190,62 @@ int M17processLICH(dsd_state * state, dsd_opts * opts, uint8_t lich_bits[96])
void M17processCodec2(dsd_opts * opts, dsd_state * state, uint8_t payload[128])
{
int i;
unsigned char bytes[16];
unsigned char bytes1[8];
unsigned char bytes2[8];
for (i = 0; i < 8; i++)
{
bytes[i+0] = (unsigned char)ConvertBitIntoBytes(&payload[i*8*1], 8);
bytes[i+8] = (unsigned char)ConvertBitIntoBytes(&payload[i*8*2], 8);
bytes1[i] = (unsigned char)ConvertBitIntoBytes(&payload[i*8*1], 8);
bytes2[i] = (unsigned char)ConvertBitIntoBytes(&payload[i*8*2], 8);
bytes1[i] = (unsigned char)ConvertBitIntoBytes(&payload[i*8+0], 8);
bytes2[i] = (unsigned char)ConvertBitIntoBytes(&payload[i*8+64], 8);
}
//TODO: A switch to flip whether or not fullrate or halfrate
//halfrate format not available yet in manual, but assuming
//it'll be either byte 1 or byte 2 when passed this far
if (opts->payload == 1)
{
fprintf (stderr, "\n CODEC2: ");
for (i = 0; i < 16; i++)
fprintf (stderr, "%02X", bytes[i]);
for (i = 0; i < 8; i++)
fprintf (stderr, "%02X", bytes1[i]);
fprintf (stderr, " (3200)");
fprintf (stderr, "\n CODEC2: ");
for (i = 0; i < 8; i++)
fprintf (stderr, "%02X", bytes2[i]);
fprintf (stderr, " (3200)");
}
//TODO: Fix Speech Output speed/rate issues and also find/figure out potential memory overflow here
//TODO: Find/figure out potential memory overflow here
#ifdef USE_CODEC2
size_t nsam;
nsam = codec2_samples_per_frame(state->codec2_3200);
//use pointer and allocate memory -- increasing allocation doesn't appear to cause any issues of note just yet
short * speech = malloc (sizeof (short) * (nsam*2)); //*2
short samp1[nsam*2];
short samp2[nsam*2];
memset (samp1, 0, sizeof(samp1));
memset (samp2, 0, sizeof(samp2));
//original -- working but speech rate in .rrc files is too fast, but sounds really clean
//may need to collect enough samples first for smooth playback, or upsample them
codec2_decode(state->codec2_3200, speech, bytes);
pa_simple_write(opts->pulse_digi_dev_out, speech, nsam*2, NULL);
//okay, I think I have fixed the speech output, it sounds natural speed now
//even for an AI voice, but still have to use the 'big chungus' memory shim until I find the issue
//testing area
// codec2_decode(state->codec2_3200, speech, bytes1);
// codec2_decode(state->codec2_3200, speech, bytes2);
// pa_simple_write(opts->pulse_digi_dev_out, speech, nsam, NULL);
// pa_simple_write(opts->pulse_digi_dev_out, speech, nsam, NULL);
codec2_decode(state->codec2_3200, samp1, bytes1);
codec2_decode(state->codec2_3200, samp2, bytes2);
//testing area
// codec2_decode(state->codec2_3200, speech, bytes1);
// pa_simple_write(opts->pulse_digi_dev_out, speech, nsam, NULL);
// codec2_decode(state->codec2_3200, speech, bytes2);
// pa_simple_write(opts->pulse_digi_dev_out, speech, nsam, NULL);
if (opts->audio_out_type == 0)
{
pa_simple_write(opts->pulse_digi_dev_out, samp1, nsam*2, NULL);
pa_simple_write(opts->pulse_digi_dev_out, samp2, nsam*2, NULL);
}
if (opts->audio_out_type == 5)
{
write (opts->audio_out_fd, samp1, nsam*2);
write (opts->audio_out_fd, samp2, nsam*2);
}
free(speech); //ba dum tiss
//TODO: Wav file saving
//TODO: Codec2 Raw file saving
#endif
@ -342,7 +349,7 @@ void M17prepareStream(dsd_opts * opts, dsd_state * state, uint8_t m17_bits[368])
else if (state->m17_str_dt == 0) fprintf (stderr, " RES;");
else fprintf (stderr, " UNK;");
if (opts->payload == 1)
if (opts->payload == 1 && state->m17_str_dt != 2)
{
fprintf (stderr, "\n STREAM: ");
for (i = 0; i < 18; i++)
@ -393,7 +400,8 @@ void processM17(dsd_opts * opts, dsd_state * state)
for (i = 0; i < 368; i++)
m17_int_bits[i] = (m17_rnd_bits[i] ^ m17_scramble[i]) & 1;
//deinterleave the bit array using Quadratic Permutation Polynomial function π(x) = (45x + 92x2 ) mod 368
//deinterleave the bit array using Quadratic Permutation Polynomial
//function π(x) = (45x + 92x2 ) mod 368
for (i = 0; i < 368; i++)
{
x = ((45*i)+(92*i*i)) % 368;
@ -409,7 +417,7 @@ void processM17(dsd_opts * opts, dsd_state * state)
if (lich_err == 0)
M17prepareStream(opts, state, m17_bits);
if (lich_err != 0) state->lastsynctype = -1;
// if (lich_err != 0) state->lastsynctype = -1; //disable
//ending linebreak
fprintf (stderr, "\n");