Rename Resampler to ResamplingQueue, to avoid conflicts with zita-resampler.

[nageru] / resampling_queue.h

diff --git a/resampling_queue.h b/resampling_queue.h
new file mode 100644 (file)
index 0000000..c085ece
--- /dev/null
+++ b/resampling_queue.h
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+#ifndef _RESAMPLING_QUEUE_H
+#define _RESAMPLING_QUEUE_H 1
+
+// 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 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
+// as a pitch shift.) Of course, the process introduces some delay; you specify a target delay
+// (typically measured in milliseconds, although more is fine) and the algorithm works to
+// provide exactly that.
+//
+// A/V sync is a much harder problem than one would intuitively assume. This implementation
+// is based on a 2012 paper by Fons Adriaensen, “Controlling adaptive resampling”
+// (http://kokkinizita.linuxaudio.org/papers/adapt-resamp.pdf). The paper gives an algorithm
+// that converges to jitter of <100 ns; the basic idea is to measure the _rate_ the input
+// queue fills and is drained (as opposed to the length of the queue itself), and smoothly
+// adjust the resampling rate so that it reaches steady state at the desired delay.
+//
+// Parts of the code is adapted from Adriaensen's project Zita-ajbridge (based on the same
+// algorithm), although it has been heavily reworked for this use case. Original copyright follows:
+//
+//  Copyright (C) 2012-2015 Fons Adriaensen <fons@linuxaudio.org>
+//    
+//  This program is free software; you can redistribute it and/or modify
+//  it under the terms of the GNU General Public License as published by
+//  the Free Software Foundation; either version 3 of the License, or
+//  (at your option) any later version.
+//
+//  This program is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+//  GNU General Public License for more details.
+//
+//  You should have received a copy of the GNU General Public License
+//  along with this program.  If not, see <http://www.gnu.org/licenses/>.
+
+#include <stdint.h>
+#include <stdlib.h>
+#include <sys/types.h>
+#include <zita-resampler/vresampler.h>
+#include <deque>
+#include <memory>
+
+class ResamplingQueue {
+public:
+       ResamplingQueue(unsigned freq_in, unsigned freq_out, unsigned num_channels = 2);
+
+       // Note: pts is always in seconds.
+       void add_input_samples(double pts, const float *samples, ssize_t num_samples);
+       bool get_output_samples(double pts, float *samples, ssize_t num_samples);  // Returns false if underrun.
+
+private:
+       void init_loop_filter(double bandwidth_hz);
+
+       VResampler vresampler;
+
+       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;
+
+       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().
+
+       ssize_t total_consumed_samples = 0;
+
+       // Duration of last input block, in seconds.
+       double last_input_len;
+
+       // Filter state for the loop filter.
+       double z1 = 0.0, z2 = 0.0, z3 = 0.0;
+
+       // Ratio between the two frequencies.
+       double ratio;
+
+       // 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 = 4800.0;
+
+       // Input samples not yet fed into the resampler.
+       // TODO: Use a circular buffer instead, for efficiency.
+       std::deque<float> buffer;
+};
+
+#endif  // !defined(_RESAMPLING_QUEUE_H)