Precalculate the VU/LRA meter images as pixmaps, since the painting is so slow.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index 861ea94d221bae78c93c06ec41f203aa30cd0905..5fa13d5cc224890a7050f04642538ad2a3e3b84d 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -157,6 +157,8 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
        // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
        // and there's a limit to how important the peak meter is.
        peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16);
+
+       alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
 }
 
 Mixer::~Mixer()
@@ -578,20 +580,7 @@ void Mixer::process_audio_one_frame()
 
 //     float limiter_att, compressor_att;
 
-       // Then a limiter at +0 dB (so, -14 dBFS) to take out the worst peaks only.
-       // Note that since ratio is not infinite, we could go slightly higher than this.
-       // Probably more tuning is warranted here.
-       if (limiter_enabled) {
-               float threshold = pow(10.0f, limiter_threshold_dbfs / 20.0f);
-               float ratio = 30.0f;
-               float attack_time = 0.0f;  // Instant.
-               float release_time = 0.005f;
-               float makeup_gain = 1.0f;  // 0 dB.
-               limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-//             limiter_att = limiter.get_attenuation();
-       }
-
-       // Finally, the real compressor.
+       // The real compressor.
        if (compressor_enabled) {
                float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
                float ratio = 20.0f;
@@ -602,6 +591,18 @@ void Mixer::process_audio_one_frame()
 //             compressor_att = compressor.get_attenuation();
        }
 
+       // 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 ratio = 30.0f;
+               float attack_time = 0.0f;  // Instant.
+               float release_time = 0.020f;
+               float makeup_gain = 1.0f;  // 0 dB.
+               limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+//             limiter_att = limiter.get_attenuation();
+       }
+
 //     printf("limiter=%+5.1f  compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
 
        // Upsample 4x to find interpolated peak.
@@ -624,7 +625,12 @@ void Mixer::process_audio_one_frame()
        float *ptrs[] = { left.data(), right.data() };
        r128.process(left.size(), ptrs);
 
-       // Actually add the samples to the output.
+       // Send the samples to the sound card.
+       if (alsa) {
+               alsa->write(samples_out);
+       }
+
+       // And finally add them to the output.
        h264_encoder->add_audio(pts_int, move(samples_out));
 }