X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.h;h=961c2a6c20948d72db4ca150aa7ec8c570d8e44d;hb=64ecf5ca155de13a76edaaefc6c4fd1f8e8f92f7;hp=8f74cc53206300ff6ee3f036dbd99859489351f4;hpb=f6724e935a578f9643a473bc99b4c3ed1e394be2;p=nageru

diff --git a/mixer.h b/mixer.h
index 8f74cc5..961c2a6 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;
@@ -168,6 +171,37 @@ public:
 		compressor_enabled = enabled;
 	}
 
+	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;
+	}
+
+	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);
+	}
+
+	void set_final_makeup_gain_auto(bool enabled)
+	{
+		std::unique_lock<std::mutex> lock(compressor_mutex);
+		final_makeup_gain_auto = enabled;
+	}
+
+	void schedule_cut()
+	{
+		should_cut = true;
+	}
+
 	void reset_meters();
 
 private:
@@ -177,7 +211,7 @@ private:
 	void place_rectangle(movit::Effect *resample_effect, movit::Effect *padding_effect, float x0, float y0, float x1, float y1);
 	void thread_func();
 	void audio_thread_func();
-	void process_audio_one_frame(int64_t frame_pts_int);
+	void process_audio_one_frame(int64_t frame_pts_int, int num_samples);
 	void subsample_chroma(GLuint src_tex, GLuint dst_dst);
 	void release_display_frame(DisplayFrame *frame);
 	double pts() { return double(pts_int) / TIMEBASE; }
@@ -209,17 +243,25 @@ private:
 		bool new_data_ready = false;  // Whether new_frame contains anything.
 		bool should_quit = false;
 		RefCountedFrame new_frame;
+		int64_t new_frame_length;  // In TIMEBASE units.
+		bool new_frame_interlaced;
+		unsigned new_frame_field;  // Which field (0 or 1) of the frame to use. Always 0 for progressive.
 		GLsync new_data_ready_fence;  // Whether new_frame is ready for rendering.
 		std::condition_variable new_data_ready_changed;  // Set whenever new_data_ready is changed.
 		unsigned dropped_frames = 0;  // Before new_frame.
 
+		// Accumulated errors in number of 1/TIMEBASE samples. If OUTPUT_FREQUENCY divided by
+		// frame rate is integer, will always stay zero.
+		unsigned fractional_samples = 0;
+
 		std::mutex audio_mutex;
 		std::unique_ptr<ResamplingQueue> resampling_queue;  // Under audio_mutex.
 		int last_timecode = -1;  // Unwrapped.
+		int64_t next_local_pts = 0;  // Beginning of next frame, in TIMEBASE units.
 	};
 	CaptureCard cards[MAX_CARDS];  // protected by <bmusb_mutex>
 
-	RefCountedFrame bmusb_current_rendering_frame[MAX_CARDS];
+	InputState input_state;
 
 	class OutputChannel {
 	public:
@@ -243,9 +285,12 @@ 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};
@@ -254,10 +299,12 @@ private:
 	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 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.
@@ -266,11 +313,18 @@ 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 {
+		int64_t pts_int;
+		int num_samples;
+	};
 	std::mutex audio_mutex;
-	std::condition_variable audio_pts_queue_changed;
-	std::queue<int64_t> audio_pts_queue;
+	std::condition_variable audio_task_queue_changed;
+	std::queue<AudioTask> audio_task_queue;  // Under audio_mutex.
 };
 
 extern Mixer *global_mixer;