diff --git a/src/dsd_audio2.c b/src/dsd_audio2.c index a1805ae..4d3c645 100644 --- a/src/dsd_audio2.c +++ b/src/dsd_audio2.c @@ -17,45 +17,45 @@ //simple method -- produces cleaner results, but can be muted (or very loud) at times //has manual control only, no auto gain -void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot) -{ - int i; - float mmax = 0.75f; - float mmin = -0.75f; - float df = 3276.7f; +// void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot) +// { +// int i; +// float mmax = 0.75f; +// float mmin = -0.75f; +// float df = 3276.7f; - //Default gain value of 1.0f on audio_gain == 0 - float gain = 1.0f; +// //Default gain value of 1.0f on audio_gain == 0 +// float gain = 1.0f; - //make it so that gain is 1.0f on 25.0f aout, and 2.0f on 50 aout - if (opts->audio_gain != 0 && slot == 0) - gain = state->aout_gain / 25.0f; +// //make it so that gain is 1.0f on 25.0f aout, and 2.0f on 50 aout +// if (opts->audio_gain != 0 && slot == 0) +// gain = state->aout_gain / 25.0f; - if (opts->audio_gain != 0 && slot == 1) - gain = state->aout_gainR / 25.0f; +// if (opts->audio_gain != 0 && slot == 1) +// gain = state->aout_gainR / 25.0f; - //mono output handles slightly different, need to further decimate - if (opts->pulse_digi_out_channels == 1) - df *= 4.0f; +// //mono output handles slightly different, need to further decimate +// if (opts->pulse_digi_out_channels == 1) +// df *= 4.0f; - for (i = 0; i < 160; i++) - { - //simple decimation - samp[i] /= df; - samp[i] *= 0.65f; +// for (i = 0; i < 160; i++) +// { +// //simple decimation +// samp[i] /= df; +// samp[i] *= 0.65f; - //simple clipping - if (samp[i] > mmax) - samp[i] = mmax; - if (samp[i] < mmin) - samp[i] = mmin; +// //simple clipping +// if (samp[i] > mmax) +// samp[i] = mmax; +// if (samp[i] < mmin) +// samp[i] = mmin; - //user gain factor - samp[i] *= gain; +// //user gain factor +// samp[i] *= gain; - } +// } -} +// } //float stereo mix 3v2 DMR void playSynthesizedVoiceFS3 (dsd_opts * opts, dsd_state * state) @@ -375,7 +375,7 @@ void playSynthesizedVoiceFS4 (dsd_opts * opts, dsd_state * state) } -//float stereo mix -- when using P25p1 and P25p2, we need to send P25p1 to this, also DMR MS/Simplex and YSF +//float stereo mix -- when using Float Stereo Output, we need to send P25p1, DMR MS/Simplex, DStar, and YSF here void playSynthesizedVoiceFS (dsd_opts * opts, dsd_state * state) { @@ -386,7 +386,7 @@ void playSynthesizedVoiceFS (dsd_opts * opts, dsd_state * state) //note, always set a stereo float mix to a baseline value and not zero! memset (stereo_samp1, 0.1f, sizeof(stereo_samp1)); - //TODO: ENC Check on P25p1, NXDN, dPMR, etc + //TODO: ENC Check on P25p1, DMR MS, etc encL = 0; //TODO: add option to bypass enc with a toggle as well @@ -460,6 +460,10 @@ void playSynthesizedVoiceFM (dsd_opts * opts, dsd_state * state) agf(opts, state, state->f_l,0); + //mono needs an additional decimation + for (int i = 0; i < 160; i++) + state->f_l[i] *= 0.5f; + if (opts->slot1_on == 0) goto vfm_end; if (opts->audio_out_type == 0) @@ -879,80 +883,82 @@ void soft_tonef (float samp[160], int n, int ID, int AD) } //older version, does better at normalizing audio, but also sounds 'flatter' and 'muddier' -//probably too much compression and adjustments on the samples +//probably too much compression and adjustments on the samples (tones have a slight tremolo effect) //Remus, enable this one and disable the one above if you prefer -// void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot) -// { -// int i, j, run; -// run = 1; -// float empty[160]; -// memset (empty, 0.1f, sizeof(empty)); +void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot) +{ + int i, j, run; + run = 1; + float empty[160]; + memset (empty, 0.1f, sizeof(empty)); -// float mmax = 0.75f; -// float mmin = -0.75f; -// float aavg = 0.0f; //average of the absolute value -// float df; //decimation value -// df = 3277.0f; //test value + float mmax = 0.90f; + float mmin = -0.90f; + float aavg = 0.0f; //average of the absolute value + float df; //decimation value + df = 3277.0f; //test value -// //trying things -// float gain = 1.0f; + //trying things + float gain = 1.0f; -// if (opts->audio_gain != 0) -// gain = opts->audio_gain / 25.0f; + if (opts->audio_gain != 0) + gain = opts->audio_gain / 25.0f; -// //this comparison is to determine whether or not to run gain on 'empty' samples (2v last 2, silent frames, etc) -// if (memcmp(empty, samp, sizeof(empty)) == 0) run = 0; -// if (run == 0) goto AGF_END; + //this comparison is to determine whether or not to run gain on 'empty' samples (2v last 2, silent frames, etc) + if (memcmp(empty, samp, sizeof(empty)) == 0) run = 0; + if (run == 0) goto AGF_END; -// for (j = 0; j < 8; j++) -// { + for (j = 0; j < 8; j++) + { -// if (slot == 0) -// df = (384.0f / (float)opts->pulse_digi_out_channels) * (50.0f - state->aout_gain); -// if (slot == 1) -// df = (384.0f / (float)opts->pulse_digi_out_channels) * (50.0f - state->aout_gainR); + if (slot == 0) + df = 384.0f * (50.0f - state->aout_gain); + if (slot == 1) + df = 384.0f * (50.0f - state->aout_gainR); -// for (i = 0; i < 20; i++) -// { + for (i = 0; i < 20; i++) + { -// samp[(j*20)+i] = samp[(j*20)+i] / df; + samp[(j*20)+i] = samp[(j*20)+i] / df; -// samp[(j*20)+i] *= gain * 0.75f; + // samp[(j*20)+i] *= gain; + // aavg += fabsf(samp[i]); -// aavg += fabsf(samp[i]); + //simple clipping + if (samp[(j*20)+i] > mmax) + samp[(j*20)+i] = mmax; + if (samp[(j*20)+i] < mmin) + samp[(j*20)+i] = mmin; -// //simple clipping -// if (samp[(j*20)+i] > mmax) -// samp[(j*20)+i] = mmax; -// if (samp[(j*20)+i] < mmin) -// samp[(j*20)+i] = mmin; + samp[(j*20)+i] *= gain * 0.8f; + aavg += fabsf(samp[i]); -// } //i loop + } //i loop -// aavg /= 20.0f; //optimal value? + aavg /= 20.0f; //average of the 20 samples -// //debug -// // fprintf (stderr, "\nS%d - DF = %f AAVG = %f", slot, df, aavg); + //debug + // fprintf (stderr, "\nS%d - DF = %f AAVG = %f", slot, df, aavg); -// if (slot == 0) -// { -// if (aavg < 0.075f && state->aout_gain < 46.0f) state->aout_gain += 0.5f; -// if (aavg >= 0.075f && state->aout_gain > 1.0f) state->aout_gain -= 0.5f; -// } + if (slot == 0) + { + if (aavg < 0.075f && state->aout_gain < 46.0f) state->aout_gain += 0.5f; + if (aavg >= 0.075f && state->aout_gain > 1.0f) state->aout_gain -= 0.5f; + } -// if (slot == 1) -// { -// if (aavg < 0.075f && state->aout_gainR < 46.0f) state->aout_gainR += 0.5f; -// if (aavg >= 0.075f && state->aout_gainR > 1.0f) state->aout_gainR -= 0.5f; -// } + if (slot == 1) + { + if (aavg < 0.075f && state->aout_gainR < 46.0f) state->aout_gainR += 0.5f; + if (aavg >= 0.075f && state->aout_gainR > 1.0f) state->aout_gainR -= 0.5f; + } -// // aavg = 0.0f; //reset? + aavg = 0.0f; //reset -// } //j loop + } //j loop -// AGF_END: ; //do nothing + AGF_END: ; //do nothing -// } +} //the previous version, not quite as good at audio gain normalization, but keeping it just in case // void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot) diff --git a/src/dsd_main.c b/src/dsd_main.c index 4d847fe..94bc017 100644 --- a/src/dsd_main.c +++ b/src/dsd_main.c @@ -1199,8 +1199,8 @@ usage () printf (" null for no audio output\n"); printf (" -d Create mbe data files, use this directory (TDMA version is experimental)\n"); printf (" -r Read/Play saved mbe data from file(s)\n"); - printf (" -g Audio Output Gain (SHORT) -- (default = 0 = auto, disable = -1)\n"); - printf (" -g Audio Manual Gain (FLOAT) -- (0 = 100%%)(Manual: 1 = 2%%; 50 = 100%%)\n"); + printf (" -g Audio Output Gain (Default: 0 = Auto; )\n"); + printf (" (Manual: 1 = 2%%; 50 = 100%%)\n"); printf (" -w Output synthesized speech to a .wav file, FDMA modes only.\n"); printf (" -7 Create/Use Custom directory for Per Call decoded .wav file saving.\n"); printf (" (Single Nested Directory Only! Use Before the -P option!)\n");