X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.h;h=ffae67a6e398687640c155ae12d8dd81c33f11e0;hb=8fe6a683cb5bc9f04555c8cb9257f33c4d356ded;hp=00c1e1519c7dddb2b648cc6ab952ee00d3e43cd6;hpb=538aa2fb443b25f34feafb4a50bb52206f7fe79c;p=nageru

diff --git a/mixer.h b/mixer.h
index 00c1e15..ffae67a 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;
@@ -59,11 +65,8 @@ 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 +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 gain_staging_db, float final_makeup_gain_db)> audio_level_callback_t;
 	void set_audio_level_callback(audio_level_callback_t callback)
 	{
 		audio_level_callback = callback;
@@ -114,12 +119,97 @@ public:
 		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;
+	}
+
+	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 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:
 	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; }
@@ -148,21 +238,28 @@ 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:
@@ -184,13 +281,47 @@ 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 compressor_mutex;
+	Ebu_r128_proc r128;  // Under compressor_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;
 
-	// TODO: Implement oversampled peak detection.
-	float peak = 0.0f;
+	// First compressor; takes us up to about -12 dBFS.
+	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;  // 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.
+	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};
+
+	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_task_queue_changed;
+	std::queue<AudioTask> audio_task_queue;  // Under audio_mutex.
 };
 
 extern Mixer *global_mixer;