Hook up the reset meters button.

[nageru] / mixer.h

diff --git a/mixer.h b/mixer.h
index bf53c2482e23362e0cba1ff589cbf72d76d78dd0..6725a7f46e242fe44bdba8a7c0e583b422b42102 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>
@@ -29,6 +30,8 @@
 #include "resampler.h"
 #include "theme.h"
 #include "timebase.h"
+#include "stereocompressor.h"
+#include "filter.h"
 
 class H264Encoder;
 class QSurface;
@@ -59,11 +62,8 @@ public:
        enum Output {
                OUTPUT_LIVE = 0,
                OUTPUT_PREVIEW,
-               OUTPUT_INPUT0,
-               OUTPUT_INPUT1,
-               OUTPUT_INPUT2,
-               OUTPUT_INPUT3,
-               NUM_OUTPUTS
+               OUTPUT_INPUT0,  // 1, 2, 3, up to 15 follow numerically.
+               NUM_OUTPUTS = 18
        };
 
        struct DisplayFrame {
@@ -98,7 +98,9 @@ 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;
@@ -109,12 +111,40 @@ public:
                return theme->get_transition_names(pts());
        }
 
+       unsigned get_num_channels() const
+       {
+               return theme->get_num_channels();
+       }
+
+       std::string get_channel_name(unsigned channel) const
+       {
+               return theme->get_channel_name(channel);
+       }
+
+       bool get_supports_set_wb(unsigned channel) const
+       {
+               return theme->get_supports_set_wb(channel);
+       }
+
+       void set_wb(unsigned channel, double r, double g, double b) const
+       {
+               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,
                FrameAllocator::Frame audio_frame, size_t audio_offset, uint16_t audio_format);
        void place_rectangle(movit::Effect *resample_effect, movit::Effect *padding_effect, float x0, float y0, float x1, float y1);
        void thread_func();
+       void process_audio_one_frame();
        void subsample_chroma(GLuint src_tex, GLuint dst_dst);
        void release_display_frame(DisplayFrame *frame);
        double pts() { return double(pts_int) / TIMEBASE; }
@@ -143,11 +173,10 @@ private:
                QSurface *surface;
                QOpenGLContext *context;
 
-               bool new_data_ready = false;  // Whether new_frame and new_frame_audio contains anything.
+               bool new_data_ready = false;  // Whether new_frame contains anything.
                bool should_quit = false;
                RefCountedFrame new_frame;
                GLsync new_data_ready_fence;  // Whether new_frame is ready for rendering.
-               std::vector<float> new_frame_audio;
                std::condition_variable new_data_ready_changed;  // Set whenever new_data_ready is changed.
                unsigned dropped_frames = 0;  // Before new_frame.
 
@@ -185,7 +214,17 @@ private:
        Ebu_r128_proc r128;
 
        // TODO: Implement oversampled peak detection.
-       float peak = 0.0f;
+       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;
+
+       StereoCompressor limiter;
+       StereoCompressor compressor;
 };
 
 extern Mixer *global_mixer;