1 // Parts of the code is adapted from Adriaensen's project Zita-ajbridge
2 // (as of November 2015), although it has been heavily reworked for this use
3 // case. Original copyright follows:
5 // Copyright (C) 2012-2015 Fons Adriaensen <fons@linuxaudio.org>
7 // This program is free software; you can redistribute it and/or modify
8 // it under the terms of the GNU General Public License as published by
9 // the Free Software Foundation; either version 3 of the License, or
10 // (at your option) any later version.
12 // This program is distributed in the hope that it will be useful,
13 // but WITHOUT ANY WARRANTY; without even the implied warranty of
14 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 // GNU General Public License for more details.
17 // You should have received a copy of the GNU General Public License
18 // along with this program. If not, see <http://www.gnu.org/licenses/>.
20 #include "resampling_queue.h"
26 #include <zita-resampler/vresampler.h>
28 ResamplingQueue::ResamplingQueue(unsigned freq_in, unsigned freq_out, unsigned num_channels)
29 : freq_in(freq_in), freq_out(freq_out), num_channels(num_channels),
30 ratio(double(freq_out) / double(freq_in))
32 vresampler.setup(ratio, num_channels, /*hlen=*/32);
34 // Prime the resampler so there's no more delay.
35 vresampler.inp_count = vresampler.inpsize() / 2 - 1;
36 vresampler.out_count = 1048576;
37 vresampler.process ();
40 void ResamplingQueue::add_input_samples(double pts, const float *samples, ssize_t num_samples)
43 // Synthesize a fake length.
44 last_input_len = double(num_samples) / freq_in;
47 last_input_len = pts - last_input_pts;
55 for (ssize_t i = 0; i < num_samples * num_channels; ++i) {
56 buffer.push_back(samples[i]);
60 bool ResamplingQueue::get_output_samples(double pts, float *samples, ssize_t num_samples)
62 double last_output_len;
64 // Synthesize a fake length.
65 last_output_len = double(num_samples) / freq_out;
67 last_output_len = pts - last_output_pts;
69 last_output_pts = pts;
71 // Using the time point since just before the last call to add_input_samples() as a base,
72 // estimate actual delay based on activity since then, measured in number of input samples:
73 double actual_delay = 0.0;
74 actual_delay += (k_a1 - k_a0) * last_output_len / last_input_len; // Inserted samples since k_a0, rescaled for the different time periods.
75 actual_delay += k_a0 - total_consumed_samples; // Samples inserted before k_a0 but not consumed yet.
76 actual_delay += vresampler.inpdist(); // Delay in the resampler itself.
77 double err = actual_delay - expected_delay;
78 if (first_output && err < 0.0) {
79 // Before the very first block, insert artificial delay based on our initial estimate,
80 // so that we don't need a long period to stabilize at the beginning.
81 int delay_samples_to_add = lrintf(-err);
82 for (ssize_t i = 0; i < delay_samples_to_add * num_channels; ++i) {
83 buffer.push_front(0.0f);
85 total_consumed_samples -= delay_samples_to_add; // Equivalent to increasing k_a0 and k_a1.
86 err += delay_samples_to_add;
90 // Compute loop filter coefficients for the two filters. We need to compute them
91 // every time, since they depend on the number of samples the user asked for.
93 // The loop bandwidth is at 0.02 Hz; we trust the initial estimate quite well,
94 // and our jitter is pretty large since none of the threads involved run at
95 // real-time priority.
96 double loop_bandwidth_hz = 0.02;
98 // Set filters. The first filter much wider than the first one (20x as wide).
99 double w = (2.0 * M_PI) * loop_bandwidth_hz * num_samples / freq_out;
100 double w0 = 1.0 - exp(-20.0 * w);
101 double w1 = w * 1.5 / num_samples / ratio;
104 // Filter <err> through the loop filter to find the correction ratio.
105 z1 += w0 * (w1 * err - z1);
106 z2 += w0 * (z1 - z2);
108 double rcorr = 1.0 - z2 - z3;
109 if (rcorr > 1.05) rcorr = 1.05;
110 if (rcorr < 0.95) rcorr = 0.95;
111 vresampler.set_rratio(rcorr);
113 // Finally actually resample, consuming exactly <num_samples> output samples.
114 vresampler.out_data = samples;
115 vresampler.out_count = num_samples;
116 while (vresampler.out_count > 0) {
117 if (buffer.empty()) {
118 // This should never happen unless delay is set way too low,
119 // or we're dropping a lot of data.
120 fprintf(stderr, "PANIC: Out of input samples to resample, still need %d output samples!\n",
121 int(vresampler.out_count));
122 memset(vresampler.out_data, 0, vresampler.out_count * 2 * sizeof(float));
127 size_t num_input_samples = sizeof(inbuf) / (sizeof(float) * num_channels);
128 if (num_input_samples * num_channels > buffer.size()) {
129 num_input_samples = buffer.size() / num_channels;
131 for (size_t i = 0; i < num_input_samples * num_channels; ++i) {
132 inbuf[i] = buffer[i];
135 vresampler.inp_count = num_input_samples;
136 vresampler.inp_data = inbuf;
138 vresampler.process();
140 size_t consumed_samples = num_input_samples - vresampler.inp_count;
141 total_consumed_samples += consumed_samples;
142 buffer.erase(buffer.begin(), buffer.begin() + consumed_samples * num_channels);