--- /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 <stdio.h>
+#include <math.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]);
+ }
+}
+
+void 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 starts at 1.0 Hz, then goes down to 0.05 Hz after four seconds.
+ double loop_bandwidth_hz = (k_a0 < 4 * freq_in) ? 1.0 : 0.05;
+
+ // Set first filter much wider than the first one (20x as wide).
+ double w = (2.0 * M_PI) * 20.0 * loop_bandwidth_hz * num_samples / freq_out;
+ double w0 = 1.0 - exp(-w);
+
+ // Set second filter.
+ w = (2.0 * M_PI) * loop_bandwidth_hz * ratio / freq_out;
+ double w1 = w * 1.6;
+ double w2 = w * num_samples / 1.6;
+
+ // 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));
+ break;
+ }
+
+ 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);
+ }
+}
projects / nageru / blobdiff