Add a beginning UI for mapping inputs. No logic connected yet.

[nageru] / audio_mixer.cpp

diff --git a/audio_mixer.cpp b/audio_mixer.cpp
index 069a810e1ad8a705431017e02ebbe2847b23a2cd..2544ad1deef84635cf5fd28d487dd78ba7293201 100644 (file)
--- a/audio_mixer.cpp
+++ b/audio_mixer.cpp
@@ -6,6 +6,7 @@
 #include <stdio.h>
 #include <cmath>
 
+#include "db.h"
 #include "flags.h"
 #include "timebase.h"
 
@@ -133,13 +134,15 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
                                num_samples,
                                rate_adjustment_policy);
                }
+
+               float volume = from_db(cards[card_index].fader_volume_db);
                if (card_index == 0) {
                        for (unsigned i = 0; i < num_samples * 2; ++i) {
-                               samples_out[i] = samples_card[i];
+                               samples_out[i] = samples_card[i] * volume;
                        }
                } else {
                        for (unsigned i = 0; i < num_samples * 2; ++i) {
-                               samples_out[i] += samples_card[i];
+                               samples_out[i] += samples_card[i] * volume;
                        }
                }
        }
@@ -167,12 +170,12 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
                                float ratio = 20.0f;
                                float attack_time = 0.5f;
                                float release_time = 20.0f;
-                               float makeup_gain = pow(10.0f, (ref_level_dbfs - (-40.0f)) / 20.0f);  // +26 dB.
+                               float makeup_gain = from_db(ref_level_dbfs - (-40.0f));  // +26 dB.
                                level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-                               gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain);
+                               gain_staging_db = to_db(level_compressor.get_attenuation() * makeup_gain);
                        } else {
                                // Just apply the gain we already had.
-                               float g = pow(10.0f, gain_staging_db / 20.0f);
+                               float g = from_db(gain_staging_db);
                                for (size_t i = 0; i < samples_out.size(); ++i) {
                                        samples_out[i] *= g;
                                }
@@ -181,16 +184,16 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
 
        #if 0
                printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
-                       level_compressor.get_level(), 20.0 * log10(level_compressor.get_level()),
-                       level_compressor.get_attenuation(), 20.0 * log10(level_compressor.get_attenuation()),
-                       20.0 * log10(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
+                       level_compressor.get_level(), to_db(level_compressor.get_level()),
+                       level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
+                       to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
        #endif
 
        //      float limiter_att, compressor_att;
 
                // The real compressor.
                if (compressor_enabled) {
-                       float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
+                       float threshold = from_db(compressor_threshold_dbfs);
                        float ratio = 20.0f;
                        float attack_time = 0.005f;
                        float release_time = 0.040f;
@@ -202,7 +205,7 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
                // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
                // Note that since ratio is not infinite, we could go slightly higher than this.
                if (limiter_enabled) {
-                       float threshold = pow(10.0f, limiter_threshold_dbfs / 20.0f);
+                       float threshold = from_db(limiter_threshold_dbfs);
                        float ratio = 30.0f;
                        float attack_time = 0.0f;  // Instant.
                        float release_time = 0.020f;
@@ -211,7 +214,7 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
        //              limiter_att = limiter.get_attenuation();
                }
 
-       //      printf("limiter=%+5.1f  compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
+       //      printf("limiter=%+5.1f  compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
        }
 
        // At this point, we are most likely close to +0 LU, but all of our
@@ -224,11 +227,11 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
        // something we get out per-sample.
        //
        // Note that there's a feedback loop here, so we choose a very slow filter
-       // (half-time of 100 seconds).
+       // (half-time of 30 seconds).
        double target_loudness_factor, alpha;
        double loudness_lu = loudness_lufs - ref_level_lufs;
-       double current_makeup_lu = 20.0f * log10(final_makeup_gain);
-       target_loudness_factor = pow(10.0f, -loudness_lu / 20.0f);
+       double current_makeup_lu = to_db(final_makeup_gain);
+       target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
 
        // If we're outside +/- 5 LU uncorrected, we don't count it as
        // a normal signal (probably silence) and don't change the
@@ -238,7 +241,7 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
        } else {
                // Formula adapted from
                // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
-               const double half_time_s = 100.0;
+               const double half_time_s = 30.0;
                const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
                alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
        }