--- /dev/null
+// Parts of the code is adapted from Adriaensen's project Zita-ajbridge
+// (as of November 2015), 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 "resampling_queue.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <zita-resampler/vresampler.h>
+#include <algorithm>
+#include <cmath>
+
+using namespace std;
+using namespace std::chrono;
+
+ResamplingQueue::ResamplingQueue(DeviceSpec device_spec, unsigned freq_in, unsigned freq_out, unsigned num_channels, double expected_delay_seconds)
+ : device_spec(device_spec), freq_in(freq_in), freq_out(freq_out), num_channels(num_channels),
+ current_estimated_freq_in(freq_in),
+ ratio(double(freq_out) / double(freq_in)), expected_delay(expected_delay_seconds * OUTPUT_FREQUENCY)
+{
+ 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 ResamplingQueue::add_input_samples(steady_clock::time_point ts, const float *samples, ssize_t num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
+{
+ if (num_samples == 0) {
+ return;
+ }
+
+ assert(duration<double>(ts.time_since_epoch()).count() >= 0.0);
+
+ bool good_sample = (rate_adjustment_policy == ADJUST_RATE);
+ if (good_sample && a1.good_sample) {
+ a0 = a1;
+ }
+ a1.ts = ts;
+ a1.input_samples_received += num_samples;
+ a1.good_sample = good_sample;
+ if (a0.good_sample && a1.good_sample) {
+ current_estimated_freq_in = (a1.input_samples_received - a0.input_samples_received) / duration<double>(a1.ts - a0.ts).count();
+ if (!(current_estimated_freq_in >= 0.0)) {
+ fprintf(stderr, "%s: PANIC: Input audio clock going backwards, ignoring.\n",
+ spec_to_string(device_spec).c_str());
+ current_estimated_freq_in = freq_in;
+ }
+
+ // Bound the frequency, so that a single wild result won't throw the filter off guard.
+ current_estimated_freq_in = min(current_estimated_freq_in, 1.2 * freq_in);
+ current_estimated_freq_in = max(current_estimated_freq_in, 0.8 * freq_in);
+ }
+
+ buffer.insert(buffer.end(), samples, samples + num_samples * num_channels);
+}
+
+bool ResamplingQueue::get_output_samples(steady_clock::time_point ts, float *samples, ssize_t num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
+{
+ assert(num_samples > 0);
+ if (a1.input_samples_received == 0) {
+ // No data yet, just return zeros.
+ memset(samples, 0, num_samples * num_channels * sizeof(float));
+ return true;
+ }
+
+ // This can happen when we get dropped frames on the master card.
+ if (duration<double>(ts.time_since_epoch()).count() <= 0.0) {
+ rate_adjustment_policy = DO_NOT_ADJUST_RATE;
+ }
+
+ if (rate_adjustment_policy == ADJUST_RATE && (a0.good_sample || a1.good_sample)) {
+ // Estimate the current number of input samples produced at
+ // this instant in time, by extrapolating from the last known
+ // good point. Note that we could be extrapolating backward or
+ // forward, depending on the timing of the calls.
+ const InputPoint &base_point = a1.good_sample ? a1 : a0;
+ assert(duration<double>(base_point.ts.time_since_epoch()).count() >= 0.0);
+
+ // NOTE: Due to extrapolation, input_samples_received can
+ // actually go negative here the few first calls (ie., we asked
+ // about a timestamp where we hadn't actually started producing
+ // samples yet), but that is harmless.
+ const double input_samples_received = base_point.input_samples_received +
+ current_estimated_freq_in * duration<double>(ts - base_point.ts).count();
+
+ // Estimate the number of input samples _consumed_ after we've run the resampler.
+ const double input_samples_consumed = total_consumed_samples +
+ num_samples / (ratio * rcorr);
+
+ double actual_delay = input_samples_received - input_samples_consumed;
+ actual_delay += vresampler.inpdist(); // Delay in the resampler itself.
+ double err = actual_delay - expected_delay;
+ if (first_output) {
+ // 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.
+ if (err < 0.0) {
+ 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 input_samples_received on a0 and a1.
+ err += delay_samples_to_add;
+ } else if (err > 0.0) {
+ int delay_samples_to_remove = min<int>(lrintf(err), buffer.size() / num_channels);
+ buffer.erase(buffer.begin(), buffer.begin() + delay_samples_to_remove * num_channels);
+ total_consumed_samples += delay_samples_to_remove;
+ err -= delay_samples_to_remove;
+ }
+ }
+ 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; our jitter is pretty large
+ // since none of the threads involved run at real-time priority.
+ // However, the first four seconds, we use a larger loop bandwidth (2 Hz),
+ // because there's a lot going on during startup, and thus the
+ // initial estimate might be tainted by jitter during that phase,
+ // and we want to converge faster.
+ //
+ // NOTE: The above logic might only hold during Nageru startup
+ // (we start ResamplingQueues also when we e.g. switch sound sources),
+ // but in general, a little bit of increased timing jitter is acceptable
+ // right after a setup change like this.
+ double loop_bandwidth_hz = (total_consumed_samples < 4 * freq_in) ? 0.2 : 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;
+ rcorr = 1.0 - z2 - z3;
+ if (rcorr > 1.05) rcorr = 1.05;
+ if (rcorr < 0.95) rcorr = 0.95;
+ assert(!isnan(rcorr));
+ vresampler.set_rratio(rcorr);
+ }
+
+ // Finally actually resample, producing 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, "%s: PANIC: Out of input samples to resample, still need %d output samples! (correction factor is %f)\n",
+ spec_to_string(device_spec).c_str(), int(vresampler.out_count), rcorr);
+ memset(vresampler.out_data, 0, vresampler.out_count * num_channels * sizeof(float));
+
+ // Reset the loop filter.
+ z1 = z2 = z3 = 0.0;
+
+ 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;
+ }
+ copy(buffer.begin(), buffer.begin() + num_input_samples * num_channels, inbuf);
+
+ vresampler.inp_count = num_input_samples;
+ vresampler.inp_data = inbuf;
+
+ int err = vresampler.process();
+ assert(err == 0);
+
+ 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