From a5db02775e318d1b51a6db5558ce2f013c74a7af Mon Sep 17 00:00:00 2001 From: Andreas Eversberg Date: Tue, 15 Mar 2016 19:26:13 +0100 Subject: [PATCH] fixup debug class definition: rename DFSK and DAUDIO to DDSP --- src/anetz/dsp.c | 20 ++++++++++---------- src/bnetz/dsp.c | 28 ++++++++++++++-------------- src/common/debug.c | 3 +-- src/common/debug.h | 15 +++++++-------- src/common/jitter.c | 2 +- src/common/loss.c | 6 +++--- src/nmt/dsp.c | 38 +++++++++++++++++++------------------- 7 files changed, 55 insertions(+), 57 deletions(-) diff --git a/src/anetz/dsp.c b/src/anetz/dsp.c index 7280515..bdc0985 100644 --- a/src/anetz/dsp.c +++ b/src/anetz/dsp.c @@ -60,7 +60,7 @@ void dsp_init(void) { int i; - PDEBUG(DFSK, DEBUG_DEBUG, "Generating sine table.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table.\n"); for (i = 0; i < 256; i++) { dsp_sine[i] = (int)(sin((double)i / 256.0 * 2.0 * PI) * TX_PEAK); } @@ -78,15 +78,15 @@ int dsp_init_sender(anetz_t *anetz) double coeff; int detect_tone = (anetz->sender.loopback) ? 0 : 1; - PDEBUG(DFSK, DEBUG_DEBUG, "Init DSP for 'Sender'.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Init DSP for 'Sender'.\n"); audio_init_loss(&anetz->sender.loss, LOSS_INTERVAL, anetz->sender.loss_volume, LOSS_TIME); anetz->samples_per_chunk = anetz->sender.samplerate * CHUNK_DURATION; - PDEBUG(DFSK, DEBUG_DEBUG, "Using %d samples per chunk duration.\n", anetz->samples_per_chunk); + PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per chunk duration.\n", anetz->samples_per_chunk); spl = calloc(1, anetz->samples_per_chunk << 1); if (!spl) { - PDEBUG(DFSK, DEBUG_ERROR, "No memory!\n"); + PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; } anetz->fsk_filter_spl = spl; @@ -95,9 +95,9 @@ int dsp_init_sender(anetz_t *anetz) coeff = 2.0 * cos(2.0 * PI * fsk_tones[detect_tone] / (double)anetz->sender.samplerate); anetz->fsk_tone_coeff = coeff * 32768.0; - PDEBUG(DFSK, DEBUG_DEBUG, "RX %.0f Hz coeff = %d\n", fsk_tones[detect_tone], (int)anetz->fsk_tone_coeff); + PDEBUG(DDSP, DEBUG_DEBUG, "RX %.0f Hz coeff = %d\n", fsk_tones[detect_tone], (int)anetz->fsk_tone_coeff); anetz->tone_phaseshift256 = 256.0 / ((double)anetz->sender.samplerate / fsk_tones[0]); - PDEBUG(DFSK, DEBUG_DEBUG, "TX %.0f Hz phaseshift = %.4f\n", fsk_tones[0], anetz->tone_phaseshift256); + PDEBUG(DDSP, DEBUG_DEBUG, "TX %.0f Hz phaseshift = %.4f\n", fsk_tones[0], anetz->tone_phaseshift256); return 0; } @@ -105,7 +105,7 @@ int dsp_init_sender(anetz_t *anetz) /* Cleanup transceiver instance. */ void dsp_cleanup_sender(anetz_t *anetz) { - PDEBUG(DFSK, DEBUG_DEBUG, "Cleanup DSP for 'Sender'.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup DSP for 'Sender'.\n"); if (anetz->fsk_filter_spl) { free(anetz->fsk_filter_spl); @@ -119,7 +119,7 @@ static void fsk_receive_tone(anetz_t *anetz, int tone, int goodtone, double leve /* lost tone because it is not good anymore or has changed */ if (!goodtone || tone != anetz->tone_detected) { if (anetz->tone_count >= TONE_DETECT_TH) { - PDEBUG(DFSK, DEBUG_DEBUG, "Lost %.0f Hz tone after %d ms.\n", fsk_tones[anetz->tone_detected], 1000.0 * CHUNK_DURATION * anetz->tone_count); + PDEBUG(DDSP, DEBUG_DEBUG, "Lost %.0f Hz tone after %d ms.\n", fsk_tones[anetz->tone_detected], 1000.0 * CHUNK_DURATION * anetz->tone_count); anetz_receive_tone(anetz, -1); } if (goodtone) @@ -136,7 +136,7 @@ static void fsk_receive_tone(anetz_t *anetz, int tone, int goodtone, double leve if (anetz->tone_count >= TONE_DETECT_TH) audio_reset_loss(&anetz->sender.loss); if (anetz->tone_count == TONE_DETECT_TH) { - PDEBUG(DFSK, DEBUG_DEBUG, "Detecting continous %.0f Hz tone. (level = %d%%)\n", fsk_tones[anetz->tone_detected], (int)(level * 100)); + PDEBUG(DDSP, DEBUG_DEBUG, "Detecting continous %.0f Hz tone. (level = %d%%)\n", fsk_tones[anetz->tone_detected], (int)(level * 100)); anetz_receive_tone(anetz, anetz->tone_detected); } } @@ -156,7 +156,7 @@ static void fsk_decode_chunk(anetz_t *anetz, int16_t *spl, int max) /* show quality of tone */ if (anetz->sender.loopback) { /* adjust level, so we get peak of sine curve */ - PDEBUG(DFSK, DEBUG_NOTICE, "Quality Tone:%3.0f%% Level:%3.0f%%\n", result / level * 100.0, level / 0.63662 * 100.0); + PDEBUG(DDSP, DEBUG_NOTICE, "Quality Tone:%3.0f%% Level:%3.0f%%\n", result / level * 100.0, level / 0.63662 * 100.0); } /* adjust level, so we get peak of sine curve */ diff --git a/src/bnetz/dsp.c b/src/bnetz/dsp.c index db339a8..9670492 100644 --- a/src/bnetz/dsp.c +++ b/src/bnetz/dsp.c @@ -58,7 +58,7 @@ void dsp_init(void) { int i; - PDEBUG(DFSK, DEBUG_DEBUG, "Generating sine table.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table.\n"); for (i = 0; i < 256; i++) { dsp_sine[i] = (int)(sin((double)i / 256.0 * 2.0 * PI) * TX_PEAK); } @@ -72,27 +72,27 @@ int dsp_init_sender(bnetz_t *bnetz) int i; if ((bnetz->sender.samplerate % 1000)) { - PDEBUG(DFSK, DEBUG_ERROR, "Samples rate must be a multiple of 1000 bits per second.\n"); + PDEBUG(DDSP, DEBUG_ERROR, "Samples rate must be a multiple of 1000 bits per second.\n"); return -EINVAL; } - PDEBUG(DFSK, DEBUG_DEBUG, "Init DSP for 'Sender'.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Init DSP for 'Sender'.\n"); audio_init_loss(&bnetz->sender.loss, LOSS_INTERVAL, bnetz->sender.loss_volume, LOSS_TIME); bnetz->samples_per_bit = bnetz->sender.samplerate * BIT_DURATION; - PDEBUG(DFSK, DEBUG_DEBUG, "Using %d samples per bit duration.\n", bnetz->samples_per_bit); + PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per bit duration.\n", bnetz->samples_per_bit); bnetz->fsk_filter_step = bnetz->sender.samplerate * FILTER_STEP; - PDEBUG(DFSK, DEBUG_DEBUG, "Using %d samples per filter step.\n", bnetz->fsk_filter_step); + PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per filter step.\n", bnetz->fsk_filter_step); spl = calloc(16, bnetz->samples_per_bit * sizeof(*spl)); if (!spl) { - PDEBUG(DFSK, DEBUG_ERROR, "No memory!\n"); + PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; } bnetz->telegramm_spl = spl; spl = calloc(1, bnetz->samples_per_bit * sizeof(*spl)); if (!spl) { - PDEBUG(DFSK, DEBUG_ERROR, "No memory!\n"); + PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; } bnetz->fsk_filter_spl = spl; @@ -104,10 +104,10 @@ int dsp_init_sender(bnetz_t *bnetz) for (i = 0; i < 2; i++) { coeff = 2.0 * cos(2.0 * PI * fsk_bits[i] / (double)bnetz->sender.samplerate); bnetz->fsk_coeff[i] = coeff * 32768.0; - PDEBUG(DFSK, DEBUG_DEBUG, "coeff[%d] = %d (must be -3601 and 2573 at 8000hz)\n", i, (int)bnetz->fsk_coeff[i]); + PDEBUG(DDSP, DEBUG_DEBUG, "coeff[%d] = %d (must be -3601 and 2573 at 8000hz)\n", i, (int)bnetz->fsk_coeff[i]); bnetz->phaseshift256[i] = 256.0 / ((double)bnetz->sender.samplerate / fsk_bits[i]); - PDEBUG(DFSK, DEBUG_DEBUG, "phaseshift[%d] = %.4f (must be arround 64 at 8000hz)\n", i, bnetz->phaseshift256[i]); + PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift[%d] = %.4f (must be arround 64 at 8000hz)\n", i, bnetz->phaseshift256[i]); } return 0; @@ -116,7 +116,7 @@ int dsp_init_sender(bnetz_t *bnetz) /* Cleanup transceiver instance. */ void dsp_cleanup_sender(bnetz_t *bnetz) { - PDEBUG(DFSK, DEBUG_DEBUG, "Cleanup DSP for 'Sender'.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup DSP for 'Sender'.\n"); if (bnetz->telegramm_spl) { free(bnetz->telegramm_spl); @@ -134,7 +134,7 @@ static void fsk_receive_tone(bnetz_t *bnetz, int bit, int goodtone, double level /* lost tone because it is not good anymore or has changed */ if (!goodtone || bit != bnetz->tone_detected) { if (bnetz->tone_count >= TONE_DETECT_TH) { - PDEBUG(DFSK, DEBUG_DEBUG, "Lost %.0f Hz tone after %d ms.\n", fsk_bits[bnetz->tone_detected], bnetz->tone_count); + PDEBUG(DDSP, DEBUG_DEBUG, "Lost %.0f Hz tone after %d ms.\n", fsk_bits[bnetz->tone_detected], bnetz->tone_count); bnetz_receive_tone(bnetz, -1); } if (goodtone) @@ -151,7 +151,7 @@ static void fsk_receive_tone(bnetz_t *bnetz, int bit, int goodtone, double level if (bnetz->tone_count >= TONE_DETECT_TH) audio_reset_loss(&bnetz->sender.loss); if (bnetz->tone_count == TONE_DETECT_TH) { - PDEBUG(DFSK, DEBUG_DEBUG, "Detecting continous %.0f Hz tone. (level = %d%%)\n", fsk_bits[bnetz->tone_detected], (int)(level * 100)); + PDEBUG(DDSP, DEBUG_DEBUG, "Detecting continous %.0f Hz tone. (level = %d%%)\n", fsk_bits[bnetz->tone_detected], (int)(level * 100)); bnetz_receive_tone(bnetz, bnetz->tone_detected); } } @@ -339,10 +339,10 @@ static int fsk_telegramm(bnetz_t *bnetz, int16_t *samples, int length) next_telegramm: if (!bnetz->telegramm) { /* request telegramm */ -// PDEBUG(DFSK, DEBUG_DEBUG, "Request new 'Telegramm'.\n"); +// PDEBUG(DDSP, DEBUG_DEBUG, "Request new 'Telegramm'.\n"); telegramm = bnetz_get_telegramm(bnetz); if (!telegramm) { - PDEBUG(DFSK, DEBUG_DEBUG, "Stop sending 'Telegramm'.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Stop sending 'Telegramm'.\n"); return length; } bnetz->telegramm = 1; diff --git a/src/common/debug.c b/src/common/debug.c index 1b733c2..49c99ec 100644 --- a/src/common/debug.c +++ b/src/common/debug.c @@ -35,8 +35,7 @@ struct debug_cat { } debug_cat[] = { { "sender", "\033[1;33m" }, { "sound", "\033[0;35m" }, - { "fsk", "\033[0;31m" }, - { "audio", "\033[0;31m" }, + { "dsp", "\033[0;31m" }, { "anetz", "\033[1;34m" }, { "bnetz", "\033[1;34m" }, { "nmt", "\033[1;34m" }, diff --git a/src/common/debug.h b/src/common/debug.h index 5989d00..deb1276 100644 --- a/src/common/debug.h +++ b/src/common/debug.h @@ -6,14 +6,13 @@ #define DSENDER 0 #define DSOUND 1 -#define DFSK 2 -#define DAUDIO 3 -#define DANETZ 4 -#define DBNETZ 5 -#define DNMT 6 -#define DFRAME 7 -#define DCALL 8 -#define DMNCC 9 +#define DDSP 2 +#define DANETZ 3 +#define DBNETZ 4 +#define DNMT 5 +#define DFRAME 6 +#define DCALL 7 +#define DMNCC 8 #define PDEBUG(cat, level, fmt, arg...) _printdebug(__FILE__, __FUNCTION__, __LINE__, cat, level, fmt, ## arg) void _printdebug(const char *file, const char *function, int line, int cat, int level, const char *fmt, ...); diff --git a/src/common/jitter.c b/src/common/jitter.c index d81fea3..2d5e901 100644 --- a/src/common/jitter.c +++ b/src/common/jitter.c @@ -31,7 +31,7 @@ int jitter_create(jitter_t *jitter, int length) memset(jitter, 0, sizeof(jitter)); jitter->spl = calloc(length * sizeof(int16_t), 1); if (!jitter->spl) { - PDEBUG(DAUDIO, DEBUG_ERROR, "No memory for jitter buffer.\n"); + PDEBUG(DDSP, DEBUG_ERROR, "No memory for jitter buffer.\n"); return -ENOMEM; } jitter->len = length; diff --git a/src/common/loss.c b/src/common/loss.c index 66a5968..d8ce2a5 100644 --- a/src/common/loss.c +++ b/src/common/loss.c @@ -44,7 +44,7 @@ void audio_init_loss(loss_t *loss, int interval, double threshold, int seconds) void audio_reset_loss(loss_t *loss) { if (loss->interval_count > 0) { - PDEBUG(DAUDIO, DEBUG_DEBUG, "Signal is recovered (loss is gone).\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Signal is recovered (loss is gone).\n"); loss->interval_count = 0; } loss->level = 0; @@ -71,7 +71,7 @@ int audio_detect_loss(loss_t *loss, double level) /* normalize level */ loss->level = loss->level / loss->level_count; - PDEBUG(DAUDIO, DEBUG_DEBUG, "Noise level = %.0f%%\n", loss->level * 100); + PDEBUG(DDSP, DEBUG_DEBUG, "Noise level = %.0f%%\n", loss->level * 100); diff = loss->level / loss->level_last; if (diff < 1.0) @@ -81,7 +81,7 @@ int audio_detect_loss(loss_t *loss, double level) loss->level_count = 0; if (diff < LOSS_MAX_DIFF && loss->level_last > loss->threshold) { loss->interval_count++; - PDEBUG(DAUDIO, DEBUG_DEBUG, "Detected signal loss %d for intervals level change %.0f%% (below %.0f%%).\n", loss->interval_count, diff * 100 - 100, LOSS_MAX_DIFF * 100 - 100); + PDEBUG(DDSP, DEBUG_DEBUG, "Detected signal loss %d for intervals level change %.0f%% (below %.0f%%).\n", loss->interval_count, diff * 100 - 100, LOSS_MAX_DIFF * 100 - 100); } else if (loss->interval_count > 0) { audio_reset_loss(loss); } diff --git a/src/nmt/dsp.c b/src/nmt/dsp.c index a5bcec6..64ff38d 100644 --- a/src/nmt/dsp.c +++ b/src/nmt/dsp.c @@ -68,7 +68,7 @@ void dsp_init(void) int i; double s; - PDEBUG(DFSK, DEBUG_DEBUG, "Generating sine table for supervisory signal.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for supervisory signal.\n"); for (i = 0; i < 256; i++) { s = sin((double)i / 256.0 * 2.0 * PI); dsp_sine_super[i] = (int)(s * TX_PEAK_SUPER); @@ -86,29 +86,29 @@ int dsp_init_sender(nmt_t *nmt) init_compander(&nmt->cstate, 8000, 3.0, 13.5); if ((nmt->sender.samplerate % (BIT_RATE * STEPS_PER_BIT))) { - PDEBUG(DFSK, DEBUG_ERROR, "Sample rate must be a multiple of %d bits per second.\n", BIT_RATE * STEPS_PER_BIT); + PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be a multiple of %d bits per second.\n", BIT_RATE * STEPS_PER_BIT); return -EINVAL; } /* this should not happen. it is implied by previous check */ if (nmt->supervisory && nmt->sender.samplerate < 12000) { - PDEBUG(DFSK, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process supervisory signal.\n"); + PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process supervisory signal.\n"); return -EINVAL; } - PDEBUG(DFSK, DEBUG_DEBUG, "Init DSP for Transceiver.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n"); /* allocate sample for 2 bits with 2 polarities */ nmt->samples_per_bit = nmt->sender.samplerate / BIT_RATE; - PDEBUG(DFSK, DEBUG_DEBUG, "Using %d samples per bit duration.\n", nmt->samples_per_bit); + PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per bit duration.\n", nmt->samples_per_bit); nmt->fsk_filter_step = nmt->samples_per_bit / STEPS_PER_BIT; - PDEBUG(DFSK, DEBUG_DEBUG, "Using %d samples per filter step.\n", nmt->fsk_filter_step); + PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per filter step.\n", nmt->fsk_filter_step); nmt->fsk_sine[0][0] = calloc(4, nmt->samples_per_bit * sizeof(int16_t)); nmt->fsk_sine[0][1] = nmt->fsk_sine[0][0] + nmt->samples_per_bit; nmt->fsk_sine[1][0] = nmt->fsk_sine[0][1] + nmt->samples_per_bit; nmt->fsk_sine[1][1] = nmt->fsk_sine[1][0] + nmt->samples_per_bit; if (!nmt->fsk_sine[0][0]) { - PDEBUG(DFSK, DEBUG_ERROR, "No memory!\n"); + PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; } @@ -123,7 +123,7 @@ int dsp_init_sender(nmt_t *nmt) /* allocate ring buffers, one bit duration */ spl = calloc(1, nmt->samples_per_bit * sizeof(*spl)); if (!spl) { - PDEBUG(DFSK, DEBUG_ERROR, "No memory!\n"); + PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; } nmt->fsk_filter_spl = spl; @@ -132,7 +132,7 @@ int dsp_init_sender(nmt_t *nmt) /* allocate transmit buffer for a complete frame */ spl = calloc(166, nmt->samples_per_bit * sizeof(*spl)); if (!spl) { - PDEBUG(DFSK, DEBUG_ERROR, "No memory!\n"); + PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; } nmt->frame_spl = spl; @@ -141,7 +141,7 @@ int dsp_init_sender(nmt_t *nmt) nmt->super_samples = (int)((double)nmt->sender.samplerate * SUPER_DURATION + 0.5); spl = calloc(166, nmt->super_samples * sizeof(*spl)); if (!spl) { - PDEBUG(DFSK, DEBUG_ERROR, "No memory!\n"); + PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; } nmt->super_filter_spl = spl; @@ -150,18 +150,18 @@ int dsp_init_sender(nmt_t *nmt) for (i = 0; i < 2; i++) { coeff = 2.0 * cos(2.0 * PI * fsk_bits[i] / (double)nmt->sender.samplerate); nmt->fsk_coeff[i] = coeff * 32768.0; - PDEBUG(DFSK, DEBUG_DEBUG, "coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]); + PDEBUG(DDSP, DEBUG_DEBUG, "coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]); } /* count supervidory tones */ for (i = 0; i < 5; i++) { coeff = 2.0 * cos(2.0 * PI * super_freq[i] / (double)nmt->sender.samplerate); nmt->super_coeff[i] = coeff * 32768.0; - PDEBUG(DFSK, DEBUG_DEBUG, "supervisory coeff[%d] = %d\n", i, (int)nmt->super_coeff[i]); + PDEBUG(DDSP, DEBUG_DEBUG, "supervisory coeff[%d] = %d\n", i, (int)nmt->super_coeff[i]); if (i < 4) { nmt->super_phaseshift256[i] = 256.0 / ((double)nmt->sender.samplerate / super_freq[i]); - PDEBUG(DFSK, DEBUG_DEBUG, "phaseshift_super[%d] = %.4f\n", i, nmt->super_phaseshift256[i]); + PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift_super[%d] = %.4f\n", i, nmt->super_phaseshift256[i]); } } super_reset(nmt); @@ -178,7 +178,7 @@ int dsp_init_sender(nmt_t *nmt) /* Cleanup transceiver instance. */ void dsp_cleanup_sender(nmt_t *nmt) { - PDEBUG(DFSK, DEBUG_DEBUG, "Cleanup DSP for 'Sender'.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup DSP for 'Sender'.\n"); if (nmt->frame_spl) { free(nmt->frame_spl); @@ -348,14 +348,14 @@ static void super_decode(nmt_t *nmt, int16_t *samples, int length) quality = 0; if (nmt->sender.loopback) - PDEBUG(DFSK, DEBUG_NOTICE, "Supervisory level %.2f%% quality %.0f%%\n", result[0] / 0.63662 * 100.0, quality * 100.0); + PDEBUG(DDSP, DEBUG_NOTICE, "Supervisory level %.2f%% quality %.0f%%\n", result[0] / 0.63662 * 100.0, quality * 100.0); if (quality > 0.5) { if (nmt->super_detected == 0) { nmt->super_detect_count++; if (nmt->super_detect_count == SUPER_DETECT_COUNT) { nmt->super_detected = 1; nmt->super_detect_count = 0; - PDEBUG(DFSK, DEBUG_DEBUG, "Supervisory signal detected with level=%.0f%%, quality=%.0f%%.\n", result[0] / 0.63662 * 100.0, quality * 100.0); + PDEBUG(DDSP, DEBUG_DEBUG, "Supervisory signal detected with level=%.0f%%, quality=%.0f%%.\n", result[0] / 0.63662 * 100.0, quality * 100.0); nmt_rx_super(nmt, 1, quality); } } else @@ -366,7 +366,7 @@ static void super_decode(nmt_t *nmt, int16_t *samples, int length) if (nmt->super_detect_count == SUPER_DETECT_COUNT) { nmt->super_detected = 0; nmt->super_detect_count = 0; - PDEBUG(DFSK, DEBUG_DEBUG, "Supervisory signal lost.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Supervisory signal lost.\n"); nmt_rx_super(nmt, 0, 0.0); } } else @@ -377,7 +377,7 @@ static void super_decode(nmt_t *nmt, int16_t *samples, int length) /* Reset supervisory detection states, so ongoing tone will be detected again. */ void super_reset(nmt_t *nmt) { - PDEBUG(DFSK, DEBUG_DEBUG, "Supervisory detector reset.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Supervisory detector reset.\n"); nmt->super_detected = 0; nmt->super_detect_count = 0; } @@ -464,7 +464,7 @@ next_frame: /* request frame */ frame = nmt_get_frame(nmt); if (!frame) { - PDEBUG(DFSK, DEBUG_DEBUG, "Stop sending frames.\n"); + PDEBUG(DDSP, DEBUG_DEBUG, "Stop sending frames.\n"); return length; } nmt->frame = 1;