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 764d7a2217fbbdfa6a7122960beb29668b5e207d..2544ad1deef84635cf5fd28d487dd78ba7293201 100644 (file)
--- a/audio_mixer.cpp
+++ b/audio_mixer.cpp
@@ -134,13 +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;
                        }
                }
        }
@@ -225,11 +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 = to_db(final_makeup_gain);
-       target_loudness_factor = from_db(-loudness_lu);
+       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
@@ -239,7 +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);
        }