]> git.sesse.net Git - nageru/blobdiff - mixer.h
Rename Resampler to ResamplingQueue, to avoid conflicts with zita-resampler.
[nageru] / mixer.h
diff --git a/mixer.h b/mixer.h
index 3e5aa68cf2c8c6e31065647436e2eec0f7faaf3c..3a19d3aaef946f6257a27a9ae060c0cf057306ba 100644 (file)
--- a/mixer.h
+++ b/mixer.h
@@ -10,6 +10,7 @@
 
 #include <movit/effect_chain.h>
 #include <movit/flat_input.h>
+#include <atomic>
 #include <condition_variable>
 #include <cstddef>
 #include <functional>
@@ -26,7 +27,7 @@
 #include "pbo_frame_allocator.h"
 #include "ref_counted_frame.h"
 #include "ref_counted_gl_sync.h"
-#include "resampler.h"
+#include "resampling_queue.h"
 #include "theme.h"
 #include "timebase.h"
 #include "stereocompressor.h"
@@ -97,7 +98,9 @@ public:
                output_channel[output].set_frame_ready_callback(callback);
        }
 
-       typedef std::function<void(float level_lufs, float peak_db, float global_level_lufs, float range_low_lufs, float range_high_lufs)> audio_level_callback_t;
+       typedef std::function<void(float level_lufs, float peak_db,
+                                  float global_level_lufs, float range_low_lufs, float range_high_lufs,
+                                  float auto_gain_staging_db)> audio_level_callback_t;
        void set_audio_level_callback(audio_level_callback_t callback)
        {
                audio_level_callback = callback;
@@ -128,6 +131,13 @@ public:
                theme->set_wb(channel, r, g, b);
        }
 
+       void set_locut_cutoff(float cutoff_hz)
+       {
+               locut_cutoff_hz = cutoff_hz;
+       }
+
+       void reset_meters();
+
 private:
        void bm_frame(unsigned card_index, uint16_t timecode,
                FrameAllocator::Frame video_frame, size_t video_offset, uint16_t video_format,
@@ -171,7 +181,7 @@ private:
                unsigned dropped_frames = 0;  // Before new_frame.
 
                std::mutex audio_mutex;
-               std::unique_ptr<Resampler> resampler;  // Under audio_mutex.
+               std::unique_ptr<ResamplingQueue> resampling_queue;  // Under audio_mutex.
                int last_timecode = -1;  // Unwrapped.
        };
        CaptureCard cards[MAX_CARDS];  // protected by <bmusb_mutex>
@@ -204,12 +214,17 @@ private:
        Ebu_r128_proc r128;
 
        // TODO: Implement oversampled peak detection.
-       float peak = 0.0f;
+       std::atomic<float> peak{0.0f};
 
-       StereoFilter locut;  // Cutoff 150 Hz, 24 dB/oct.
+       StereoFilter locut;  // Default cutoff 150 Hz, 24 dB/oct.
+       std::atomic<float> locut_cutoff_hz;
 
        // First compressor; takes us up to about -12 dBFS.
        StereoCompressor level_compressor;
+       float last_gain_staging_db = 0.0f;
+
+       StereoCompressor limiter;
+       StereoCompressor compressor;
 };
 
 extern Mixer *global_mixer;