]> git.sesse.net Git - nageru/blobdiff - mixer.h
Add a soundcard output via ALSA.
[nageru] / mixer.h
diff --git a/mixer.h b/mixer.h
index cdc6f1c8c32762b6e68f86ea61c1d3745e355f92..89a80dd52fabaccd8e81f1743c491f41b4d329ff 100644 (file)
--- a/mixer.h
+++ b/mixer.h
@@ -10,6 +10,8 @@
 
 #include <movit/effect_chain.h>
 #include <movit/flat_input.h>
+#include <zita-resampler/resampler.h>
+#include <atomic>
 #include <condition_variable>
 #include <cstddef>
 #include <functional>
 #include <vector>
 
 #include "bmusb/bmusb.h"
+#include "alsa_output.h"
 #include "ebu_r128_proc.h"
 #include "h264encode.h"
 #include "httpd.h"
 #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"
+#include "filter.h"
 
 class H264Encoder;
 class QSurface;
@@ -96,7 +100,9 @@ public:
                output_channel[output].set_frame_ready_callback(callback);
        }
 
-       typedef std::function<void(float, float, float, float, float)> 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;
@@ -127,6 +133,43 @@ public:
                theme->set_wb(channel, r, g, b);
        }
 
+       void set_locut_cutoff(float cutoff_hz)
+       {
+               locut_cutoff_hz = cutoff_hz;
+       }
+
+       float get_limiter_threshold_dbfs()
+       {
+               return limiter_threshold_dbfs;
+       }
+
+       float get_compressor_threshold_dbfs()
+       {
+               return compressor_threshold_dbfs;
+       }
+
+       void set_limiter_threshold_dbfs(float threshold_dbfs)
+       {
+               limiter_threshold_dbfs = threshold_dbfs;
+       }
+
+       void set_compressor_threshold_dbfs(float threshold_dbfs)
+       {
+               compressor_threshold_dbfs = threshold_dbfs;
+       }
+
+       void set_limiter_enabled(bool enabled)
+       {
+               limiter_enabled = enabled;
+       }
+
+       void set_compressor_enabled(bool enabled)
+       {
+               compressor_enabled = enabled;
+       }
+
+       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,
@@ -170,7 +213,7 @@ 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>
@@ -202,11 +245,26 @@ private:
        audio_level_callback_t audio_level_callback = nullptr;
        Ebu_r128_proc r128;
 
-       // TODO: Implement oversampled peak detection.
-       float peak = 0.0f;
+       Resampler peak_resampler;
+       std::atomic<float> peak{0.0f};
+
+       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;
+
+       static constexpr float ref_level_dbfs = -14.0f;
+
+       StereoCompressor limiter;
+       std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f};   // 4 dB.
+       std::atomic<bool> limiter_enabled{true};
+       StereoCompressor compressor;
+       std::atomic<float> compressor_threshold_dbfs{ref_level_dbfs - 12.0f};  // -12 dB.
+       std::atomic<bool> compressor_enabled{true};
+
+       std::unique_ptr<ALSAOutput> alsa;
 };
 
 extern Mixer *global_mixer;