mirror of
https://github.com/lwvmobile/dsd-fme.git
synced 2026-11-12 16:57:23 -05:00
More Tweaking on AGF and Float Audio;
This commit is contained in:
lwvmobile
176
src/dsd_audio2.c
176
src/dsd_audio2.c
@@ -17,45 +17,45 @@
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//simple method -- produces cleaner results, but can be muted (or very loud) at times
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//has manual control only, no auto gain
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void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot)
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{
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int i;
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float mmax = 0.75f;
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float mmin = -0.75f;
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float df = 3276.7f;
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// void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot)
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// {
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// int i;
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// float mmax = 0.75f;
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// float mmin = -0.75f;
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// float df = 3276.7f;
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//Default gain value of 1.0f on audio_gain == 0
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float gain = 1.0f;
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// //Default gain value of 1.0f on audio_gain == 0
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// float gain = 1.0f;
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//make it so that gain is 1.0f on 25.0f aout, and 2.0f on 50 aout
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if (opts->audio_gain != 0 && slot == 0)
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gain = state->aout_gain / 25.0f;
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// //make it so that gain is 1.0f on 25.0f aout, and 2.0f on 50 aout
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// if (opts->audio_gain != 0 && slot == 0)
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// gain = state->aout_gain / 25.0f;
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if (opts->audio_gain != 0 && slot == 1)
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gain = state->aout_gainR / 25.0f;
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// if (opts->audio_gain != 0 && slot == 1)
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// gain = state->aout_gainR / 25.0f;
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//mono output handles slightly different, need to further decimate
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if (opts->pulse_digi_out_channels == 1)
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df *= 4.0f;
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// //mono output handles slightly different, need to further decimate
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// if (opts->pulse_digi_out_channels == 1)
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// df *= 4.0f;
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for (i = 0; i < 160; i++)
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{
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//simple decimation
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samp[i] /= df;
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samp[i] *= 0.65f;
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// for (i = 0; i < 160; i++)
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// {
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// //simple decimation
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// samp[i] /= df;
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// samp[i] *= 0.65f;
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//simple clipping
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if (samp[i] > mmax)
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samp[i] = mmax;
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if (samp[i] < mmin)
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samp[i] = mmin;
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// //simple clipping
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// if (samp[i] > mmax)
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// samp[i] = mmax;
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// if (samp[i] < mmin)
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// samp[i] = mmin;
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//user gain factor
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samp[i] *= gain;
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// //user gain factor
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// samp[i] *= gain;
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}
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// }
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}
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// }
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//float stereo mix 3v2 DMR
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void playSynthesizedVoiceFS3 (dsd_opts * opts, dsd_state * state)
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@@ -375,7 +375,7 @@ void playSynthesizedVoiceFS4 (dsd_opts * opts, dsd_state * state)
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}
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//float stereo mix -- when using P25p1 and P25p2, we need to send P25p1 to this, also DMR MS/Simplex and YSF
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//float stereo mix -- when using Float Stereo Output, we need to send P25p1, DMR MS/Simplex, DStar, and YSF here
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void playSynthesizedVoiceFS (dsd_opts * opts, dsd_state * state)
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{
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@@ -386,7 +386,7 @@ void playSynthesizedVoiceFS (dsd_opts * opts, dsd_state * state)
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//note, always set a stereo float mix to a baseline value and not zero!
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memset (stereo_samp1, 0.1f, sizeof(stereo_samp1));
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//TODO: ENC Check on P25p1, NXDN, dPMR, etc
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//TODO: ENC Check on P25p1, DMR MS, etc
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encL = 0;
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//TODO: add option to bypass enc with a toggle as well
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@@ -460,6 +460,10 @@ void playSynthesizedVoiceFM (dsd_opts * opts, dsd_state * state)
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agf(opts, state, state->f_l,0);
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//mono needs an additional decimation
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for (int i = 0; i < 160; i++)
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state->f_l[i] *= 0.5f;
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if (opts->slot1_on == 0) goto vfm_end;
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if (opts->audio_out_type == 0)
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@@ -879,80 +883,82 @@ void soft_tonef (float samp[160], int n, int ID, int AD)
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}
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//older version, does better at normalizing audio, but also sounds 'flatter' and 'muddier'
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//probably too much compression and adjustments on the samples
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//probably too much compression and adjustments on the samples (tones have a slight tremolo effect)
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//Remus, enable this one and disable the one above if you prefer
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// void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot)
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// {
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// int i, j, run;
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// run = 1;
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// float empty[160];
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// memset (empty, 0.1f, sizeof(empty));
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void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot)
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{
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int i, j, run;
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run = 1;
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float empty[160];
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memset (empty, 0.1f, sizeof(empty));
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// float mmax = 0.75f;
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// float mmin = -0.75f;
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// float aavg = 0.0f; //average of the absolute value
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// float df; //decimation value
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// df = 3277.0f; //test value
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float mmax = 0.90f;
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float mmin = -0.90f;
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float aavg = 0.0f; //average of the absolute value
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float df; //decimation value
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df = 3277.0f; //test value
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// //trying things
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// float gain = 1.0f;
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//trying things
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float gain = 1.0f;
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// if (opts->audio_gain != 0)
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// gain = opts->audio_gain / 25.0f;
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if (opts->audio_gain != 0)
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gain = opts->audio_gain / 25.0f;
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// //this comparison is to determine whether or not to run gain on 'empty' samples (2v last 2, silent frames, etc)
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// if (memcmp(empty, samp, sizeof(empty)) == 0) run = 0;
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// if (run == 0) goto AGF_END;
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//this comparison is to determine whether or not to run gain on 'empty' samples (2v last 2, silent frames, etc)
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if (memcmp(empty, samp, sizeof(empty)) == 0) run = 0;
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if (run == 0) goto AGF_END;
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// for (j = 0; j < 8; j++)
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// {
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for (j = 0; j < 8; j++)
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{
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// if (slot == 0)
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// df = (384.0f / (float)opts->pulse_digi_out_channels) * (50.0f - state->aout_gain);
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// if (slot == 1)
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// df = (384.0f / (float)opts->pulse_digi_out_channels) * (50.0f - state->aout_gainR);
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if (slot == 0)
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df = 384.0f * (50.0f - state->aout_gain);
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if (slot == 1)
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df = 384.0f * (50.0f - state->aout_gainR);
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// for (i = 0; i < 20; i++)
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// {
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for (i = 0; i < 20; i++)
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{
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// samp[(j*20)+i] = samp[(j*20)+i] / df;
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samp[(j*20)+i] = samp[(j*20)+i] / df;
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// samp[(j*20)+i] *= gain * 0.75f;
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// samp[(j*20)+i] *= gain;
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// aavg += fabsf(samp[i]);
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// aavg += fabsf(samp[i]);
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//simple clipping
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if (samp[(j*20)+i] > mmax)
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samp[(j*20)+i] = mmax;
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if (samp[(j*20)+i] < mmin)
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samp[(j*20)+i] = mmin;
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// //simple clipping
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// if (samp[(j*20)+i] > mmax)
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// samp[(j*20)+i] = mmax;
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// if (samp[(j*20)+i] < mmin)
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// samp[(j*20)+i] = mmin;
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samp[(j*20)+i] *= gain * 0.8f;
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aavg += fabsf(samp[i]);
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// } //i loop
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} //i loop
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// aavg /= 20.0f; //optimal value?
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aavg /= 20.0f; //average of the 20 samples
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// //debug
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// // fprintf (stderr, "\nS%d - DF = %f AAVG = %f", slot, df, aavg);
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//debug
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// fprintf (stderr, "\nS%d - DF = %f AAVG = %f", slot, df, aavg);
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// if (slot == 0)
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// {
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// if (aavg < 0.075f && state->aout_gain < 46.0f) state->aout_gain += 0.5f;
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// if (aavg >= 0.075f && state->aout_gain > 1.0f) state->aout_gain -= 0.5f;
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// }
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if (slot == 0)
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{
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if (aavg < 0.075f && state->aout_gain < 46.0f) state->aout_gain += 0.5f;
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if (aavg >= 0.075f && state->aout_gain > 1.0f) state->aout_gain -= 0.5f;
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}
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// if (slot == 1)
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// {
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// if (aavg < 0.075f && state->aout_gainR < 46.0f) state->aout_gainR += 0.5f;
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// if (aavg >= 0.075f && state->aout_gainR > 1.0f) state->aout_gainR -= 0.5f;
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// }
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if (slot == 1)
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{
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if (aavg < 0.075f && state->aout_gainR < 46.0f) state->aout_gainR += 0.5f;
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if (aavg >= 0.075f && state->aout_gainR > 1.0f) state->aout_gainR -= 0.5f;
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}
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// // aavg = 0.0f; //reset?
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aavg = 0.0f; //reset
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// } //j loop
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} //j loop
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// AGF_END: ; //do nothing
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AGF_END: ; //do nothing
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// }
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}
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//the previous version, not quite as good at audio gain normalization, but keeping it just in case
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// void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot)
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@@ -1199,8 +1199,8 @@ usage ()
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printf (" null for no audio output\n");
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printf (" -d <dir> Create mbe data files, use this directory (TDMA version is experimental)\n");
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printf (" -r <files> Read/Play saved mbe data from file(s)\n");
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printf (" -g <float> Audio Output Gain (SHORT) -- (default = 0 = auto, disable = -1)\n");
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printf (" -g <float> Audio Manual Gain (FLOAT) -- (0 = 100%%)(Manual: 1 = 2%%; 50 = 100%%)\n");
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printf (" -g <float> Audio Output Gain (Default: 0 = Auto; )\n");
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printf (" (Manual: 1 = 2%%; 50 = 100%%)\n");
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printf (" -w <file> Output synthesized speech to a .wav file, FDMA modes only.\n");
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printf (" -7 <dir> Create/Use Custom directory for Per Call decoded .wav file saving.\n");
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printf (" (Single Nested Directory Only! Use Before the -P option!)\n");
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