-// 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:
+// 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>
//
#include "resampling_queue.h"
-#include <math.h>
-#include <stddef.h>
+#include <assert.h>
#include <stdio.h>
+#include <stdlib.h>
#include <string.h>
#include <zita-resampler/vresampler.h>
+#include <algorithm>
+#include <cmath>
-ResamplingQueue::ResamplingQueue(unsigned freq_in, unsigned freq_out, unsigned num_channels)
- : freq_in(freq_in), freq_out(freq_out), num_channels(num_channels),
+using namespace std;
+
+ResamplingQueue::ResamplingQueue(unsigned card_num, unsigned freq_in, unsigned freq_out, unsigned num_channels)
+ : card_num(card_num), 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);
void ResamplingQueue::add_input_samples(double pts, const float *samples, ssize_t num_samples)
{
+ if (num_samples == 0) {
+ return;
+ }
if (first_input) {
// Synthesize a fake length.
last_input_len = double(num_samples) / freq_in;
k_a0 = k_a1;
k_a1 += num_samples;
- for (ssize_t i = 0; i < num_samples * num_channels; ++i) {
- buffer.push_back(samples[i]);
- }
+ buffer.insert(buffer.end(), samples, samples + num_samples * num_channels);
}
-bool ResamplingQueue::get_output_samples(double pts, float *samples, ssize_t num_samples)
+bool ResamplingQueue::get_output_samples(double pts, float *samples, ssize_t num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
{
- 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;
+ assert(num_samples > 0);
+ if (first_input) {
+ // No data yet, just return zeros.
+ memset(samples, 0, num_samples * num_channels * sizeof(float));
+ return true;
}
- 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);
+
+ double rcorr = -1.0;
+ if (rate_adjustment_policy == ADJUST_RATE) {
+ 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;
+ assert(last_input_len != 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;
}
- 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);
+ // 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;
+ 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);
+ } else {
+ assert(rate_adjustment_policy == DO_NOT_ADJUST_RATE);
+ };
// Finally actually resample, consuming exactly <num_samples> output samples.
vresampler.out_data = samples;
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));
+ fprintf(stderr, "Card %u: PANIC: Out of input samples to resample, still need %d output samples! (correction factor is %f)\n",
+ card_num, int(vresampler.out_count), rcorr);
+ memset(vresampler.out_data, 0, vresampler.out_count * num_channels * sizeof(float));
return false;
}
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];
- }
+ copy(buffer.begin(), buffer.begin() + num_input_samples * num_channels, inbuf);
vresampler.inp_count = num_input_samples;
vresampler.inp_data = inbuf;
- vresampler.process();
+ int err = vresampler.process();
+ assert(err == 0);
size_t consumed_samples = num_input_samples - vresampler.inp_count;
total_consumed_samples += consumed_samples;
projects / nageru / blobdiff