+++ /dev/null
-#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 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
-// 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 <sys/types.h>
-#include <zita-resampler/vresampler.h>
-#include <chrono>
-#include <deque>
-#include <memory>
-
-#include "defs.h"
-#include "input_mapping.h"
-
-class ResamplingQueue {
-public:
- // 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
- // dropped frames.
- enum RateAdjustmentPolicy {
- DO_NOT_ADJUST_RATE,
- ADJUST_RATE
- };
-
- 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(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;
-
- DeviceSpec device_spec;
- unsigned freq_in, freq_out, num_channels;
-
- bool first_output = true;
-
- struct InputPoint {
- // Equivalent to t_a0 or t_a1 in the paper.
- std::chrono::steady_clock::time_point ts;
-
- // 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;
-
- // 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.
- 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.
- const double expected_delay;
-
- // 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)
projects / nageru / blobdiff