X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=resampling_queue.h;h=f0e2499cbf3300e9b4340430b85bf33f25862006;hb=4a0187ffb4075b4d217b8d9e9c96cac548b199d8;hp=662a837262e8b84fcf3680ceae65bdf38f0e0871;hpb=605ebde8d784157641733f8eddf31012227888ce;p=nageru

diff --git a/resampling_queue.h b/resampling_queue.h
index 662a837..f0e2499 100644
--- a/resampling_queue.h
+++ b/resampling_queue.h
@@ -4,8 +4,8 @@
 // Takes in samples from an input source, possibly with jitter, and outputs a fixed number
 // of samples every iteration. Used to a) change sample rates if needed, and b) deal with
 // input sources that don't have audio locked to video. For every input video
-// frame, you call add_input_samples() with the pts (measured in seconds) of the video frame,
-// taken to be the start point of the frame's audio. When you want to _output_ a finished
+// frame, you call add_input_samples() with the received time point of the video frame,
+// taken to be the _end_ point of the frame's audio. When you want to _output_ a finished
 // frame with audio, you get_output_samples() with the number of samples you want, and will
 // get exactly that number of samples back. If the input and output clocks are not in sync,
 // the audio will be stretched for you. (If they are _very_ out of sync, this will come through
@@ -40,15 +40,17 @@
 
 #include <sys/types.h>
 #include <zita-resampler/vresampler.h>
+#include <chrono>
 #include <deque>
 #include <memory>
 
 #include "defs.h"
+#include "input_mapping.h"
 
 class ResamplingQueue {
 public:
-	// card_num is for debugging outputs only.
-	ResamplingQueue(unsigned card_num, unsigned freq_in, unsigned freq_out, unsigned num_channels, double expected_delay_seconds);
+	// device_spec is for debugging outputs only.
+	ResamplingQueue(DeviceSpec device_spec, unsigned freq_in, unsigned freq_out, unsigned num_channels, double expected_delay_seconds);
 
 	// If policy is DO_NOT_ADJUST_RATE, the resampling rate will not be changed.
 	// This is primarily useful if you have an extraordinary situation, such as
@@ -58,41 +60,55 @@ public:
 		ADJUST_RATE
 	};
 
-	// Note: pts is always in seconds.
-	void add_input_samples(double pts, const float *samples, ssize_t num_samples);
+	void add_input_samples(std::chrono::steady_clock::time_point ts, const float *samples, ssize_t num_samples, RateAdjustmentPolicy rate_adjustment_policy);
 	// Returns false if underrun.
-	bool get_output_samples(double pts, float *samples, ssize_t num_samples, RateAdjustmentPolicy rate_adjustment_policy);
+	bool get_output_samples(std::chrono::steady_clock::time_point ts, float *samples, ssize_t num_samples, RateAdjustmentPolicy rate_adjustment_policy);
 
 private:
 	void init_loop_filter(double bandwidth_hz);
 
 	VResampler vresampler;
 
-	unsigned card_num;
+	DeviceSpec device_spec;
 	unsigned freq_in, freq_out, num_channels;
 
-	bool first_input = true, first_output = true;
-	double last_input_pts;   // Start of last input block, in seconds.
-	double last_output_pts;
+	bool first_output = true;
 
-	ssize_t k_a0 = 0;  // Total amount of samples inserted _before_ the last call to add_input_samples().
-	ssize_t k_a1 = 0;  // Total amount of samples inserted _after_ the last call to add_input_samples().
+	struct InputPoint {
+		// Equivalent to t_a0 or t_a1 in the paper.
+		std::chrono::steady_clock::time_point ts;
 
-	ssize_t total_consumed_samples = 0;
+		// Number of samples that have been written to the queue (in total)
+		// at this time point. Equivalent to k_a0 or k_a1 in the paper.
+		size_t input_samples_received = 0;
+
+		// Set to false if we should not use the timestamp from this sample
+		// (e.g. if it is from a dropped frame and thus bad). In particular,
+		// we will not use it for updateing current_estimated_freq_in.
+		bool good_sample = false;
+	};
+	InputPoint a0, a1;
 
-	// Duration of last input block, in seconds.
-	double last_input_len;
+	// The current rate at which we seem to get input samples, in Hz.
+	// For an ideal input, identical to freq_in.
+	double current_estimated_freq_in;
+
+	ssize_t total_consumed_samples = 0;
 
 	// Filter state for the loop filter.
 	double z1 = 0.0, z2 = 0.0, z3 = 0.0;
 
 	// Ratio between the two frequencies.
-	double ratio;
+	const double ratio;
+
+	// Current correction ratio. ratio * rcorr gives the true ratio,
+	// so values above 1.0 means to pitch down (consume input samples slower).
+	double rcorr = 1.0;
 
 	// How much delay we are expected to have, in input samples.
 	// If actual delay drifts too much away from this, we will start
 	// changing the resampling ratio to compensate.
-	double expected_delay;
+	const double expected_delay;
 
 	// Input samples not yet fed into the resampler.
 	// TODO: Use a circular buffer instead, for efficiency.