Add global DC-Filter and remove all individual DC-Filters
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71e556e7ff
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@ -450,7 +450,7 @@ int amps_create(int channel, enum amps_chan_type chan_type, const char *audiodev
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}
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/* init audio processing */
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rc = dsp_init_sender(amps, (de_emphasis == 0), tolerant);
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rc = dsp_init_sender(amps, tolerant);
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if (rc < 0) {
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PDEBUG(DAMPS, DEBUG_ERROR, "Failed to init audio processing!\n");
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goto error;
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@ -178,12 +178,11 @@ static void dsp_init_ramp(amps_t *amps)
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static void sat_reset(amps_t *amps, const char *reason);
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/* Init FSK of transceiver */
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int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
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int dsp_init_sender(amps_t *amps, int tolerant)
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{
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sample_t *spl;
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int i;
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int rc;
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double RC, dt;
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int half;
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/* attack (3ms) and recovery time (13.5ms) according to amps specs */
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@ -256,14 +255,6 @@ int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
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amps->test_phaseshift256 = 256.0 / ((double)amps->sender.samplerate / 1000.0);
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PDEBUG(DDSP, DEBUG_DEBUG, "test_phaseshift256 = %.4f\n", amps->test_phaseshift256);
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/* use this filter to remove dc level for 0-crossing detection
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* if we have de-emphasis, we don't need it, so high_pass is not set. */
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if (high_pass) {
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RC = 1.0 / (CUT_OFF_HIGHPASS * 2.0 *3.14);
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dt = 1.0 / amps->sender.samplerate;
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amps->highpass_factor = RC / (RC + dt);
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}
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/* be more tolerant when syncing */
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amps->fsk_rx_sync_tolerant = tolerant;
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@ -808,8 +799,7 @@ static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
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int max, pos;
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int i;
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/* SAT detection */
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/* SAT / signalling tone detection */
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max = amps->sat_samples;
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spl = amps->sat_filter_spl;
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pos = amps->sat_filter_pos;
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@ -853,25 +843,10 @@ static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
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void sender_receive(sender_t *sender, sample_t *samples, int length)
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{
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amps_t *amps = (amps_t *) sender;
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double x, y, x_last, y_last, factor;
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int i;
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/* high pass filter to remove 0-level
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* if factor is not set, we should already have 0-level. */
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factor = amps->highpass_factor;
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if (factor) {
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x_last = amps->highpass_x_last;
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y_last = amps->highpass_y_last;
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for (i = 0; i < length; i++) {
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x = (double)samples[i];
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y = factor * (y_last + x - x_last);
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x_last = x;
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y_last = y;
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samples[i] = y;
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}
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amps->highpass_x_last = x_last;
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amps->highpass_y_last = y_last;
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}
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/* dc filter required for FSK decoding and tone detection */
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if (amps->de_emphasis)
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dc_filter(&s->estate, samples, length);
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switch (amps->dsp_mode) {
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case DSP_MODE_OFF:
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@ -1,6 +1,6 @@
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void dsp_init(void);
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int dsp_init_sender(amps_t *amps, int high_pass, int tolerant);
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int dsp_init_sender(amps_t *amps, int tolerant);
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void dsp_cleanup_sender(amps_t *amps);
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void amps_set_dsp_mode(amps_t *amps, enum dsp_mode mode, int frame_length);
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@ -812,6 +812,8 @@ void unshrink_speech(cnetz_t *cnetz, sample_t *speech_buffer, int count)
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/* 4. de-emphasis is done by cnetz code, not by common code */
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/* de-emphasis is only used when scrambler is off, see FTZ 171 TR 60 Clause 4 */
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if (cnetz->de_emphasis)
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dc_filter(&cnetz->estate, speech_buffer, count);
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if (cnetz->de_emphasis && !cnetz->scrambler)
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de_emphasis(&cnetz->estate, speech_buffer, count);
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/* 3. descramble */
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@ -27,7 +27,7 @@
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#define PI M_PI
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#define CUT_OFF_H 300.0 /* cut-off frequency for high-pass filters */
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#define CUT_OFF_H 100.0 /* cut-off frequency for high-pass filter */
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static void gen_sine(double *samples, int num, int samplerate, double freq)
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{
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@ -106,8 +106,6 @@ void de_emphasis(emphasis_t *state, double *samples, int num)
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double x, y, y_last, factor, amp;
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int i;
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filter_process(&state->d.hp, samples, num);
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y_last = state->d.y_last;
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factor = state->d.factor;
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amp = state->d.amp;
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@ -126,3 +124,9 @@ void de_emphasis(emphasis_t *state, double *samples, int num)
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state->d.y_last = y_last;
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}
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/* high pass filter to remove DC and low frequencies */
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void dc_filter(emphasis_t *state, double *samples, int num)
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{
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filter_process(&state->d.hp, samples, num);
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}
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@ -17,4 +17,5 @@ typedef struct emphasis {
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int init_emphasis(emphasis_t *state, int samplerate, double cut_off);
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void pre_emphasis(emphasis_t *state, double *samples, int num);
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void de_emphasis(emphasis_t *state, double *samples, int num);
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void dc_filter(emphasis_t *state, double *samples, int num);
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@ -30,30 +30,26 @@
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* audio level calculation
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*/
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/* return average value (rectified value), that can be 0..1 */
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/* Return average value (rectified value)
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* The input must not have any dc offset!
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* For a perfect rectangualr wave, the result would equal the peak level.
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* For a sine wave the result would be factor (2 / PI) below peak level.
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*/
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double audio_level(sample_t *samples, int length)
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{
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double bias;
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double level;
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int sk;
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double level, sk;
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int n;
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/* calculate bias */
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bias = 0;
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for (n = 0; n < length; n++)
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bias += samples[n];
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bias = bias / length;
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/* level calculation */
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level = 0;
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for (n = 0; n < length; n++) {
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sk = samples[n] - bias;
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sk = samples[n];
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if (sk < 0)
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level -= (double)sk;
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if (sk > 0)
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level += (double)sk;
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}
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level = level / (double)length / 32767.0;
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level = level / (double)length;
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return level;
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}
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@ -79,17 +75,10 @@ void audio_goertzel_init(goertzel_t *goertzel, double freq, int samplerate)
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*/
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void audio_goertzel(goertzel_t *goertzel, sample_t *samples, int length, int offset, double *result, int k)
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{
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double bias;
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double sk, sk1, sk2;
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double cos2pik;
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int i, n;
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/* calculate bias to remove DC */
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bias = 0;
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for (n = 0; n < length; n++)
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bias += samples[n];
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bias = bias / length;
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/* we do goertzel */
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for (i = 0; i < k; i++) {
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sk = 0;
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@ -98,7 +87,7 @@ void audio_goertzel(goertzel_t *goertzel, sample_t *samples, int length, int off
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cos2pik = goertzel[i].coeff;
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/* note: after 'length' cycles, offset is restored to its initial value */
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for (n = 0; n < length; n++) {
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sk = (cos2pik * sk1) - sk2 + samples[offset++] - bias;
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sk = (cos2pik * sk1) - sk2 + samples[offset++];
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sk2 = sk1;
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sk1 = sk;
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if (offset == length)
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@ -281,7 +281,7 @@ cant_recover:
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display_wave(inst, samples[i], count);
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sender_receive(inst, samples[i], count);
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}
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/* do pre emphasis towards radio, not wave_write */
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/* do pre emphasis towards radio */
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if (inst->pre_emphasis)
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pre_emphasis(&inst->estate, samples[i], count);
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/* set paging signal */
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@ -331,9 +331,11 @@ transmit_later:
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/* rx gain */
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if (inst->rx_gain != 1.0)
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gain_samples(samples[i], count, inst->rx_gain);
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/* do de emphasis from radio (then write_wave/wave_read), receive audio, process echo test */
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if (inst->de_emphasis)
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/* do filter and de-emphasis from radio receive audio, process echo test */
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if (inst->de_emphasis) {
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dc_filter(&inst->estate, samples[i], count);
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de_emphasis(&inst->estate, samples[i], count);
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}
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if (inst->loopback != 1) {
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display_wave(inst, samples[i], count);
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sender_receive(inst, samples[i], count);
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@ -64,6 +64,7 @@ int main(void)
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for (i = 31.25; i < 4001; i = i * sqrt(sqrt(2.0))) {
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gen_samples(samples, (double)i);
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dc_filter(&estate, samples, SAMPLERATE);
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de_emphasis(&estate, samples, SAMPLERATE);
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level = get_level(samples);
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printf("%s%.0f Hz: %.1f dB", debug_db(level), i, level2db(level));
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