X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.h;h=32e8168f64cc96a62c3cdc58a51d4773417c7e48;hb=f245cb0d0453e0e8bd5c7d40720bd0a5b50454ee;hp=b9acfc4c607e2a9b880ef8dea5bc7867da9fa286;hpb=86f32b5a80b232e6d824776a87ecc1b5f8fbfa96;p=nageru

diff --git a/mixer.h b/mixer.h
index b9acfc4..32e8168 100644
--- a/mixer.h
+++ b/mixer.h
@@ -34,6 +34,8 @@
 #include "timebase.h"
 #include "stereocompressor.h"
 #include "filter.h"
+#include "input_state.h"
+#include "correlation_measurer.h"
 
 class H264Encoder;
 class QSurface;
@@ -102,7 +104,8 @@ public:
 
 	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;
+	                           float gain_staging_db, float final_makeup_gain_db,
+	                           float correlation)> audio_level_callback_t;
 	void set_audio_level_callback(audio_level_callback_t callback)
 	{
 		audio_level_callback = callback;
@@ -123,6 +126,21 @@ public:
 		return theme->get_channel_name(channel);
 	}
 
+	int get_channel_signal(unsigned channel) const
+	{
+		return theme->get_channel_signal(channel);
+	}
+
+	int map_signal(unsigned channel)
+	{
+		return theme->map_signal(channel);
+	}
+
+	void set_signal_mapping(int signal, int card)
+	{
+		return theme->set_signal_mapping(signal, card);
+	}
+
 	bool get_supports_set_wb(unsigned channel) const
 	{
 		return theme->get_supports_set_wb(channel);
@@ -138,6 +156,11 @@ public:
 		locut_cutoff_hz = cutoff_hz;
 	}
 
+	void set_locut_enabled(bool enabled)
+	{
+		locut_enabled = enabled;
+	}
+
 	float get_limiter_threshold_dbfs()
 	{
 		return limiter_threshold_dbfs;
@@ -168,16 +191,44 @@ public:
 		compressor_enabled = enabled;
 	}
 
-	void reset_meters();
+	void set_gain_staging_db(float gain_db)
+	{
+		std::unique_lock<std::mutex> lock(compressor_mutex);
+		level_compressor_enabled = false;
+		gain_staging_db = gain_db;
+	}
 
-	struct BufferedFrame {
-		RefCountedFrame frame;
-		unsigned field_number;
-	};
+	void set_gain_staging_auto(bool enabled)
+	{
+		std::unique_lock<std::mutex> lock(compressor_mutex);
+		level_compressor_enabled = enabled;
+	}
+
+	void set_final_makeup_gain_db(float gain_db)
+	{
+		std::unique_lock<std::mutex> lock(compressor_mutex);
+		final_makeup_gain_auto = false;
+		final_makeup_gain = pow(10.0f, gain_db / 20.0f);
+	}
 
-	BufferedFrame get_buffered_frame(int card, int history_pos)
+	void set_final_makeup_gain_auto(bool enabled)
 	{
-		return buffered_frames[card][history_pos];
+		std::unique_lock<std::mutex> lock(compressor_mutex);
+		final_makeup_gain_auto = enabled;
+	}
+
+	void schedule_cut()
+	{
+		should_cut = true;
+	}
+
+	void reset_meters();
+
+	unsigned get_num_cards() const { return num_cards; }
+
+	std::string get_card_description(unsigned card_index) const {
+		assert(card_index < num_cards);
+		return cards[card_index].usb->get_description();
 	}
 
 private:
@@ -237,12 +288,7 @@ private:
 	};
 	CaptureCard cards[MAX_CARDS];  // protected by <bmusb_mutex>
 
-	// For each card, the last three frames (or fields), with 0 being the
-	// most recent one. Note that we only need the actual history if we have
-	// interlaced output (for deinterlacing), so if we detect progressive input,
-	// we immediately clear out all history and all entries will point to the same
-	// frame.
-	BufferedFrame buffered_frames[MAX_CARDS][FRAME_HISTORY_LENGTH];
+	InputState input_state;
 
 	class OutputChannel {
 	public:
@@ -266,21 +312,27 @@ private:
 	std::thread mixer_thread;
 	std::thread audio_thread;
 	std::atomic<bool> should_quit{false};
+	std::atomic<bool> should_cut{false};
 
 	audio_level_callback_t audio_level_callback = nullptr;
-	Ebu_r128_proc r128;
+	std::mutex compressor_mutex;
+	Ebu_r128_proc r128;  // Under compressor_mutex.
+	CorrelationMeasurer correlation;  // Under compressor_mutex.
 
 	Resampler peak_resampler;
 	std::atomic<float> peak{0.0f};
 
-	StereoFilter locut;  // Default cutoff 150 Hz, 24 dB/oct.
+	StereoFilter locut;  // Default cutoff 120 Hz, 24 dB/oct.
 	std::atomic<float> locut_cutoff_hz;
+	std::atomic<bool> locut_enabled{true};
 
 	// First compressor; takes us up to about -12 dBFS.
-	StereoCompressor level_compressor;
-	float last_gain_staging_db = 0.0f;
+	StereoCompressor level_compressor;  // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
+	float gain_staging_db = 0.0f;  // Under compressor_mutex.
+	bool level_compressor_enabled = true;  // Under compressor_mutex.
 
-	static constexpr float ref_level_dbfs = -14.0f;
+	static constexpr float ref_level_dbfs = -14.0f;  // Chosen so that we end up around 0 LU in practice.
+	static constexpr float ref_level_lufs = -23.0f;  // 0 LU, more or less by definition.
 
 	StereoCompressor limiter;
 	std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f};   // 4 dB.
@@ -289,6 +341,9 @@ private:
 	std::atomic<float> compressor_threshold_dbfs{ref_level_dbfs - 12.0f};  // -12 dB.
 	std::atomic<bool> compressor_enabled{true};
 
+	double final_makeup_gain = 1.0;  // Under compressor_mutex. Read/write by the user. Note: Not in dB, we want the numeric precision so that we can change it slowly.
+	bool final_makeup_gain_auto = true;  // Under compressor_mutex.
+
 	std::unique_ptr<ALSAOutput> alsa;
 
 	struct AudioTask {