X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.h;h=b9acfc4c607e2a9b880ef8dea5bc7867da9fa286;hb=2464152a8c084f39b52e063da888a6a0b7ea9306;hp=64a74e2f13d855d0030a19645cdadf95956b9558;hpb=e8918c22dd8cd509d9679d50bfc6bba86661ebb3;p=nageru

diff --git a/mixer.h b/mixer.h
index 64a74e2..b9acfc4 100644
--- a/mixer.h
+++ b/mixer.h
@@ -22,6 +22,7 @@
 #include <vector>
 
 #include "bmusb/bmusb.h"
+#include "alsa_output.h"
 #include "ebu_r128_proc.h"
 #include "h264encode.h"
 #include "httpd.h"
@@ -169,13 +170,24 @@ public:
 
 	void reset_meters();
 
+	struct BufferedFrame {
+		RefCountedFrame frame;
+		unsigned field_number;
+	};
+
+	BufferedFrame get_buffered_frame(int card, int history_pos)
+	{
+		return buffered_frames[card][history_pos];
+	}
+
 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 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; }
@@ -207,17 +219,30 @@ 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];
+	// 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];
 
 	class OutputChannel {
 	public:
@@ -239,7 +264,8 @@ private:
 	OutputChannel output_channel[NUM_OUTPUTS];
 
 	std::thread mixer_thread;
-	bool should_quit = false;
+	std::thread audio_thread;
+	std::atomic<bool> should_quit{false};
 
 	audio_level_callback_t audio_level_callback = nullptr;
 	Ebu_r128_proc r128;
@@ -257,11 +283,21 @@ private:
 	static constexpr float ref_level_dbfs = -14.0f;
 
 	StereoCompressor limiter;
-	std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 0.0f};   // 0 dB.
+	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;