X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.h;h=db6db15155bacb85fd0fe011296cf770422039f5;hb=906cab7a1020e9f416a8b768c28ea25a9caac66a;hp=02c06edfa46f2d8ee12278dee03320efe979e673;hpb=9f1e8fb59e1b68b68b4bb1a05e1f4ee37ea47471;p=nageru

diff --git a/mixer.h b/mixer.h
index 02c06ed..db6db15 100644
--- 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>
@@ -20,15 +22,19 @@
 #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"
+#include "input_state.h"
 
 class H264Encoder;
 class QSurface;
@@ -95,7 +101,9 @@ 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;
@@ -126,12 +134,56 @@ 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 schedule_cut()
+	{
+		should_cut = true;
+	}
+
+	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 audio_thread_func();
+	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; }
@@ -160,21 +212,28 @@ 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;
+		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::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.
 
+		// 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<Resampler> resampler;  // Under 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:
@@ -196,13 +255,42 @@ private:
 	OutputChannel output_channel[NUM_OUTPUTS];
 
 	std::thread mixer_thread;
-	bool should_quit = false;
+	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 r128_mutex;
+	Ebu_r128_proc r128;  // Under r128_mutex.
+
+	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;
 
-	// TODO: Implement oversampled peak detection.
-	float peak = 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;
+
+	struct AudioTask {
+		int64_t pts_int;
+		int num_samples;
+	};
+	std::mutex audio_mutex;
+	std::condition_variable audio_task_queue_changed;
+	std::queue<AudioTask> audio_task_queue;  // Under audio_mutex.
 };
 
 extern Mixer *global_mixer;