+++ /dev/null
-// Parts of the code is adapted from Adriaensen's project Zita-ajbridge,
-// 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 "resampler.h"
-
-#include <math.h>
-#include <stddef.h>
-#include <stdio.h>
-#include <string.h>
-#include <zita-resampler/vresampler.h>
-
-Resampler::Resampler(unsigned freq_in, unsigned freq_out, unsigned num_channels)
- : freq_in(freq_in), freq_out(freq_out), num_channels(num_channels),
- ratio(double(freq_out) / double(freq_in))
-{
- vresampler.setup(ratio, num_channels, /*hlen=*/32);
-
- // Prime the resampler so there's no more delay.
- vresampler.inp_count = vresampler.inpsize() / 2 - 1;
- vresampler.out_count = 1048576;
- vresampler.process ();
-}
-
-void Resampler::add_input_samples(double pts, const float *samples, ssize_t num_samples)
-{
- if (first_input) {
- // Synthesize a fake length.
- last_input_len = double(num_samples) / freq_in;
- first_input = false;
- } else {
- last_input_len = pts - last_input_pts;
- }
-
- last_input_pts = pts;
-
- k_a0 = k_a1;
- k_a1 += num_samples;
-
- for (ssize_t i = 0; i < num_samples * num_channels; ++i) {
- buffer.push_back(samples[i]);
- }
-}
-
-bool Resampler::get_output_samples(double pts, float *samples, ssize_t num_samples)
-{
- double last_output_len;
- if (first_output) {
- // Synthesize a fake length.
- last_output_len = double(num_samples) / freq_out;
- } else {
- last_output_len = pts - last_output_pts;
- }
- last_output_pts = pts;
-
- // Using the time point since just before the last call to add_input_samples() as a base,
- // estimate actual delay based on activity since then, measured in number of input samples:
- double actual_delay = 0.0;
- actual_delay += (k_a1 - k_a0) * last_output_len / last_input_len; // Inserted samples since k_a0, rescaled for the different time periods.
- actual_delay += k_a0 - total_consumed_samples; // Samples inserted before k_a0 but not consumed yet.
- actual_delay += vresampler.inpdist(); // Delay in the resampler itself.
- double err = actual_delay - expected_delay;
- if (first_output && err < 0.0) {
- // Before the very first block, insert artificial delay based on our initial estimate,
- // so that we don't need a long period to stabilize at the beginning.
- int delay_samples_to_add = lrintf(-err);
- for (ssize_t i = 0; i < delay_samples_to_add * num_channels; ++i) {
- buffer.push_front(0.0f);
- }
- total_consumed_samples -= delay_samples_to_add; // Equivalent to increasing k_a0 and k_a1.
- err += delay_samples_to_add;
- first_output = false;
- }
-
- // Compute loop filter coefficients for the two filters. We need to compute them
- // every time, since they depend on the number of samples the user asked for.
- //
- // The loop bandwidth is at 0.02 Hz; we trust the initial estimate quite well,
- // and our jitter is pretty large since none of the threads involved run at
- // real-time priority.
- double loop_bandwidth_hz = 0.02;
-
- // Set filters. The first filter much wider than the first one (20x as wide).
- double w = (2.0 * M_PI) * loop_bandwidth_hz * num_samples / freq_out;
- double w0 = 1.0 - exp(-20.0 * w);
- double w1 = w * 1.5 / num_samples / ratio;
- double w2 = w / 1.5;
-
- // Filter <err> through the loop filter to find the correction ratio.
- z1 += w0 * (w1 * err - z1);
- z2 += w0 * (z1 - z2);
- z3 += w2 * z2;
- double rcorr = 1.0 - z2 - z3;
- if (rcorr > 1.05) rcorr = 1.05;
- if (rcorr < 0.95) rcorr = 0.95;
- vresampler.set_rratio(rcorr);
-
- // Finally actually resample, consuming exactly <num_samples> output samples.
- vresampler.out_data = samples;
- vresampler.out_count = num_samples;
- while (vresampler.out_count > 0) {
- if (buffer.empty()) {
- // This should never happen unless delay is set way too low,
- // or we're dropping a lot of data.
- fprintf(stderr, "PANIC: Out of input samples to resample, still need %d output samples!\n",
- int(vresampler.out_count));
- memset(vresampler.out_data, 0, vresampler.out_count * sizeof(float));
- return false;
- }
-
- float inbuf[1024];
- size_t num_input_samples = sizeof(inbuf) / (sizeof(float) * num_channels);
- if (num_input_samples * num_channels > buffer.size()) {
- num_input_samples = buffer.size() / num_channels;
- }
- for (size_t i = 0; i < num_input_samples * num_channels; ++i) {
- inbuf[i] = buffer[i];
- }
-
- vresampler.inp_count = num_input_samples;
- vresampler.inp_data = inbuf;
-
- vresampler.process();
-
- size_t consumed_samples = num_input_samples - vresampler.inp_count;
- total_consumed_samples += consumed_samples;
- buffer.erase(buffer.begin(), buffer.begin() + consumed_samples * num_channels);
- }
- return true;
-}
projects / nageru / blobdiff