tor-browser

test_resampler.cpp (53031B)

---

/*
* Copyright © 2016 Mozilla Foundation
*
* This program is made available under an ISC-style license.  See the
* accompanying file LICENSE for details.
*/
#ifndef NOMINMAX
#define NOMINMAX
#endif // NOMINMAX
#include "cubeb/cubeb.h"
#include "cubeb_audio_dump.h"
#include "cubeb_log.h"
#include "cubeb_resampler.h"
// #define ENABLE_NORMAL_LOG
// #define ENABLE_VERBOSE_LOG
#include "common.h"
#include "cubeb_resampler_internal.h"
#include "gtest/gtest.h"
#include <algorithm>
#include <cmath>
#include <iostream>
#include <queue>
#include <stdio.h>
#include <thread>

/* Windows cmath USE_MATH_DEFINE thing... */
const float PI = 3.14159265359f;

/* Testing all sample rates is very long, so if THOROUGH_TESTING is not defined,
* only part of the test suite is ran. */
#ifdef THOROUGH_TESTING
/* Some standard sample rates we're testing with. */
const uint32_t sample_rates[] = {8000,  16000, 32000, 44100,
                                48000, 88200, 96000, 192000};
/* The maximum number of channels we're resampling. */
const uint32_t max_channels = 2;
/* The minimum an maximum number of milliseconds we're resampling for. This is
* used to simulate the fact that the audio stream is resampled in chunks,
* because audio is delivered using callbacks. */
const uint32_t min_chunks = 10; /* ms */
const uint32_t max_chunks = 30; /* ms */
const uint32_t chunk_increment = 1;

#else

const uint32_t sample_rates[] = {
   8000,
   44100,
   48000,
};
const uint32_t max_channels = 2;
const uint32_t min_chunks = 10; /* ms */
const uint32_t max_chunks = 30; /* ms */
const uint32_t chunk_increment = 10;
#endif

// #define DUMP_ARRAYS
#ifdef DUMP_ARRAYS
/**
* Files produced by dump(...) can be converted to .wave files using:
*
* sox -c <channel_count> -r <rate> -e float -b 32  file.raw file.wav
*
* for floating-point audio, or:
*
* sox -c <channel_count> -r <rate> -e unsigned -b 16  file.raw file.wav
*
* for 16bit integer audio.
*/

/* Use the correct implementation of fopen, depending on the platform. */
void
fopen_portable(FILE ** f, const char * name, const char * mode)
{
#ifdef WIN32
 fopen_s(f, name, mode);
#else
 *f = fopen(name, mode);
#endif
}

template <typename T>
void
dump(const char * name, T * frames, size_t count)
{
 FILE * file;
 fopen_portable(&file, name, "wb");

 if (!file) {
   fprintf(stderr, "error opening %s\n", name);
   return;
 }

 if (count != fwrite(frames, sizeof(T), count, file)) {
   fprintf(stderr, "error writing to %s\n", name);
 }
 fclose(file);
}
#else
template <typename T>
void
dump(const char * name, T * frames, size_t count)
{
}
#endif

// The more the ratio is far from 1, the more we accept a big error.
float
epsilon_tweak_ratio(float ratio)
{
 return ratio >= 1 ? ratio : 1 / ratio;
}

// Epsilon values for comparing resampled data to expected data.
// The bigger the resampling ratio is, the more lax we are about errors.
template <typename T>
T
epsilon(float ratio);

template <>
float
epsilon(float ratio)
{
 return 0.08f * epsilon_tweak_ratio(ratio);
}

template <>
int16_t
epsilon(float ratio)
{
 return static_cast<int16_t>(10 * epsilon_tweak_ratio(ratio));
}

void
test_delay_lines(uint32_t delay_frames, uint32_t channels, uint32_t chunk_ms)
{
 const size_t length_s = 2;
 const size_t rate = 44100;
 const size_t length_frames = rate * length_s;
 delay_line<float> delay(delay_frames, channels, rate);
 auto_array<float> input;
 auto_array<float> output;
 uint32_t chunk_length = channels * chunk_ms * rate / 1000;
 uint32_t output_offset = 0;
 uint32_t channel = 0;

 /** Generate diracs every 100 frames, and check they are delayed. */
 input.push_silence(length_frames * channels);
 for (uint32_t i = 0; i < input.length() - 1; i += 100) {
   input.data()[i + channel] = 0.5;
   channel = (channel + 1) % channels;
 }
 dump("input.raw", input.data(), input.length());
 while (input.length()) {
   uint32_t to_pop =
       std::min<uint32_t>(input.length(), chunk_length * channels);
   float * in = delay.input_buffer(to_pop / channels);
   input.pop(in, to_pop);
   delay.written(to_pop / channels);
   output.push_silence(to_pop);
   delay.output(output.data() + output_offset, to_pop / channels);
   output_offset += to_pop;
 }

 // Check the diracs have been shifted by `delay_frames` frames.
 for (uint32_t i = 0; i < output.length() - delay_frames * channels + 1;
      i += 100) {
   ASSERT_EQ(output.data()[i + channel + delay_frames * channels], 0.5);
   channel = (channel + 1) % channels;
 }

 dump("output.raw", output.data(), output.length());
}
/**
* This takes sine waves with a certain `channels` count, `source_rate`, and
* resample them, by chunk of `chunk_duration` milliseconds, to `target_rate`.
* Then a sample-wise comparison is performed against a sine wave generated at
* the correct rate.
*/
template <typename T>
void
test_resampler_one_way(uint32_t channels, uint32_t source_rate,
                      uint32_t target_rate, float chunk_duration)
{
 size_t chunk_duration_in_source_frames =
     static_cast<uint32_t>(ceil(chunk_duration * source_rate / 1000.));
 float resampling_ratio = static_cast<float>(source_rate) / target_rate;
 cubeb_resampler_speex_one_way<T> resampler(channels, source_rate, target_rate,
                                            3);
 auto_array<T> source(channels * source_rate * 10);
 auto_array<T> destination(channels * target_rate * 10);
 auto_array<T> expected(channels * target_rate * 10);
 uint32_t phase_index = 0;
 uint32_t offset = 0;
 const uint32_t buf_len = 2; /* seconds */

 // generate a sine wave in each channel, at the source sample rate
 source.push_silence(channels * source_rate * buf_len);
 while (offset != source.length()) {
   float p = phase_index++ / static_cast<float>(source_rate);
   for (uint32_t j = 0; j < channels; j++) {
     source.data()[offset++] = 0.5 * sin(440. * 2 * PI * p);
   }
 }

 dump("input.raw", source.data(), source.length());

 expected.push_silence(channels * target_rate * buf_len);
 // generate a sine wave in each channel, at the target sample rate.
 // Insert silent samples at the beginning to account for the resampler
 // latency.
 offset = resampler.latency() * channels;
 for (uint32_t i = 0; i < offset; i++) {
   expected.data()[i] = 0.0f;
 }
 phase_index = 0;
 while (offset != expected.length()) {
   float p = phase_index++ / static_cast<float>(target_rate);
   for (uint32_t j = 0; j < channels; j++) {
     expected.data()[offset++] = 0.5 * sin(440. * 2 * PI * p);
   }
 }

 dump("expected.raw", expected.data(), expected.length());

 // resample by chunk
 uint32_t write_offset = 0;
 destination.push_silence(channels * target_rate * buf_len);
 while (write_offset < destination.length()) {
   size_t output_frames = static_cast<uint32_t>(
       floor(chunk_duration_in_source_frames / resampling_ratio));
   uint32_t input_frames = resampler.input_needed_for_output(output_frames);
   resampler.input(source.data(), input_frames);
   source.pop(nullptr, input_frames * channels);
   resampler.output(
       destination.data() + write_offset,
       std::min(output_frames,
                (destination.length() - write_offset) / channels));
   write_offset += output_frames * channels;
 }

 dump("output.raw", destination.data(), expected.length());

 // compare, taking the latency into account
 bool fuzzy_equal = true;
 for (uint32_t i = resampler.latency() + 1; i < expected.length(); i++) {
   float diff = fabs(expected.data()[i] - destination.data()[i]);
   if (diff > epsilon<T>(resampling_ratio)) {
     fprintf(stderr, "divergence at %d: %f %f (delta %f)\n", i,
             expected.data()[i], destination.data()[i], diff);
     fuzzy_equal = false;
   }
 }
 ASSERT_TRUE(fuzzy_equal);
}

template <typename T>
cubeb_sample_format
cubeb_format();

template <>
cubeb_sample_format
cubeb_format<float>()
{
 return CUBEB_SAMPLE_FLOAT32NE;
}

template <>
cubeb_sample_format
cubeb_format<short>()
{
 return CUBEB_SAMPLE_S16NE;
}

struct osc_state {
 osc_state()
     : input_phase_index(0), output_phase_index(0), output_offset(0),
       input_channels(0), output_channels(0)
 {
 }
 uint32_t input_phase_index;
 uint32_t max_output_phase_index;
 uint32_t output_phase_index;
 uint32_t output_offset;
 uint32_t input_channels;
 uint32_t output_channels;
 uint32_t output_rate;
 uint32_t target_rate;
 auto_array<float> input;
 auto_array<float> output;
};

uint32_t
fill_with_sine(float * buf, uint32_t rate, uint32_t channels, uint32_t frames,
              uint32_t initial_phase)
{
 uint32_t offset = 0;
 for (uint32_t i = 0; i < frames; i++) {
   float p = initial_phase++ / static_cast<float>(rate);
   for (uint32_t j = 0; j < channels; j++) {
     buf[offset++] = 0.5 * sin(440. * 2 * PI * p);
   }
 }
 return initial_phase;
}

long
data_cb_resampler(cubeb_stream * /*stm*/, void * user_ptr,
                 const void * input_buffer, void * output_buffer,
                 long frame_count)
{
 osc_state * state = reinterpret_cast<osc_state *>(user_ptr);
 const float * in = reinterpret_cast<const float *>(input_buffer);
 float * out = reinterpret_cast<float *>(output_buffer);

 state->input.push(in, frame_count * state->input_channels);

 /* Check how much output frames we need to write */
 uint32_t remaining =
     state->max_output_phase_index - state->output_phase_index;
 uint32_t to_write = std::min<uint32_t>(remaining, frame_count);
 state->output_phase_index =
     fill_with_sine(out, state->target_rate, state->output_channels, to_write,
                    state->output_phase_index);

 return to_write;
}

template <typename T>
bool
array_fuzzy_equal(const auto_array<T> & lhs, const auto_array<T> & rhs, T epsi)
{
 uint32_t len = std::min(lhs.length(), rhs.length());

 for (uint32_t i = 0; i < len; i++) {
   if (fabs(lhs.at(i) - rhs.at(i)) > epsi) {
     std::cout << "not fuzzy equal at index: " << i << " lhs: " << lhs.at(i)
               << " rhs: " << rhs.at(i)
               << " delta: " << fabs(lhs.at(i) - rhs.at(i))
               << " epsilon: " << epsi << std::endl;
     return false;
   }
 }
 return true;
}

template <typename T>
void
test_resampler_duplex(uint32_t input_channels, uint32_t output_channels,
                     uint32_t input_rate, uint32_t output_rate,
                     uint32_t target_rate, float chunk_duration)
{
 cubeb_stream_params input_params;
 cubeb_stream_params output_params;
 osc_state state;

 input_params.format = output_params.format = cubeb_format<T>();
 state.input_channels = input_params.channels = input_channels;
 state.output_channels = output_params.channels = output_channels;
 input_params.rate = input_rate;
 state.output_rate = output_params.rate = output_rate;
 state.target_rate = target_rate;
 input_params.prefs = output_params.prefs = CUBEB_STREAM_PREF_NONE;
 long got;

 cubeb_resampler * resampler = cubeb_resampler_create(
     (cubeb_stream *)nullptr, &input_params, &output_params, target_rate,
     data_cb_resampler, (void *)&state, CUBEB_RESAMPLER_QUALITY_VOIP,
     CUBEB_RESAMPLER_RECLOCK_NONE);

 long latency = cubeb_resampler_latency(resampler);

 const uint32_t duration_s = 2;
 int32_t duration_frames = duration_s * target_rate;
 uint32_t input_array_frame_count =
     ceil(chunk_duration * input_rate / 1000) +
     ceilf(static_cast<float>(input_rate) / target_rate) * 2;
 uint32_t output_array_frame_count = chunk_duration * output_rate / 1000;
 auto_array<float> input_buffer(input_channels * input_array_frame_count);
 auto_array<float> output_buffer(output_channels * output_array_frame_count);
 auto_array<float> expected_resampled_input(input_channels * duration_frames);
 auto_array<float> expected_resampled_output(output_channels * output_rate *
                                             duration_s);

 state.max_output_phase_index = duration_s * target_rate;

 expected_resampled_input.push_silence(input_channels * duration_frames);
 expected_resampled_output.push_silence(output_channels * output_rate *
                                        duration_s);

 /* expected output is a 440Hz sine wave at 16kHz */
 fill_with_sine(expected_resampled_input.data() + latency, target_rate,
                input_channels, duration_frames - latency, 0);
 /* expected output is a 440Hz sine wave at 32kHz */
 fill_with_sine(expected_resampled_output.data() + latency, output_rate,
                output_channels, output_rate * duration_s - latency, 0);

 while (state.output_phase_index != state.max_output_phase_index) {
   uint32_t leftover_samples = input_buffer.length() * input_channels;
   input_buffer.reserve(input_array_frame_count);
   state.input_phase_index = fill_with_sine(
       input_buffer.data() + leftover_samples, input_rate, input_channels,
       input_array_frame_count - leftover_samples, state.input_phase_index);
   long input_consumed = input_array_frame_count;
   input_buffer.set_length(input_array_frame_count);

   got = cubeb_resampler_fill(resampler, input_buffer.data(), &input_consumed,
                              output_buffer.data(), output_array_frame_count);

   /* handle leftover input */
   if (input_array_frame_count != static_cast<uint32_t>(input_consumed)) {
     input_buffer.pop(nullptr, input_consumed * input_channels);
   } else {
     input_buffer.clear();
   }

   state.output.push(output_buffer.data(), got * state.output_channels);
 }

 dump("input_expected.raw", expected_resampled_input.data(),
      expected_resampled_input.length());
 dump("output_expected.raw", expected_resampled_output.data(),
      expected_resampled_output.length());
 dump("input.raw", state.input.data(), state.input.length());
 dump("output.raw", state.output.data(), state.output.length());

 // This is disabled because the latency estimation in the resampler code is
 // slightly off so we can generate expected vectors.
 // See https://github.com/kinetiknz/cubeb/issues/93
 // ASSERT_TRUE(array_fuzzy_equal(state.input, expected_resampled_input,
 // epsilon<T>(input_rate/target_rate)));
 // ASSERT_TRUE(array_fuzzy_equal(state.output, expected_resampled_output,
 // epsilon<T>(output_rate/target_rate)));

 cubeb_resampler_destroy(resampler);
}

#define array_size(x) (sizeof(x) / sizeof(x[0]))

TEST(cubeb, resampler_one_way)
{
 /* Test one way resamplers */
 for (uint32_t channels = 1; channels <= max_channels; channels++) {
   for (uint32_t source_rate = 0; source_rate < array_size(sample_rates);
        source_rate++) {
     for (uint32_t dest_rate = 0; dest_rate < array_size(sample_rates);
          dest_rate++) {
       for (uint32_t chunk_duration = min_chunks; chunk_duration < max_chunks;
            chunk_duration += chunk_increment) {
         fprintf(stderr,
                 "one_way: channels: %d, source_rate: %d, dest_rate: %d, "
                 "chunk_duration: %d\n",
                 channels, sample_rates[source_rate], sample_rates[dest_rate],
                 chunk_duration);
         test_resampler_one_way<float>(channels, sample_rates[source_rate],
                                       sample_rates[dest_rate],
                                       chunk_duration);
       }
     }
   }
 }
}

TEST(cubeb, DISABLED_resampler_duplex)
{
 for (uint32_t input_channels = 1; input_channels <= max_channels;
      input_channels++) {
   for (uint32_t output_channels = 1; output_channels <= max_channels;
        output_channels++) {
     for (uint32_t source_rate_input = 0;
          source_rate_input < array_size(sample_rates); source_rate_input++) {
       for (uint32_t source_rate_output = 0;
            source_rate_output < array_size(sample_rates);
            source_rate_output++) {
         for (uint32_t dest_rate = 0; dest_rate < array_size(sample_rates);
              dest_rate++) {
           for (uint32_t chunk_duration = min_chunks;
                chunk_duration < max_chunks;
                chunk_duration += chunk_increment) {
             fprintf(stderr,
                     "input channels:%d output_channels:%d input_rate:%d "
                     "output_rate:%d target_rate:%d chunk_ms:%d\n",
                     input_channels, output_channels,
                     sample_rates[source_rate_input],
                     sample_rates[source_rate_output], sample_rates[dest_rate],
                     chunk_duration);
             test_resampler_duplex<float>(input_channels, output_channels,
                                          sample_rates[source_rate_input],
                                          sample_rates[source_rate_output],
                                          sample_rates[dest_rate],
                                          chunk_duration);
           }
         }
       }
     }
   }
 }
}

TEST(cubeb, resampler_delay_line)
{
 for (uint32_t channel = 1; channel <= 2; channel++) {
   for (uint32_t delay_frames = 4; delay_frames <= 40;
        delay_frames += chunk_increment) {
     for (uint32_t chunk_size = 10; chunk_size <= 30; chunk_size++) {
       fprintf(stderr, "channel: %d, delay_frames: %d, chunk_size: %d\n",
               channel, delay_frames, chunk_size);
       test_delay_lines(delay_frames, channel, chunk_size);
     }
   }
 }
}

long
test_output_only_noop_data_cb(cubeb_stream * /*stm*/, void * /*user_ptr*/,
                             const void * input_buffer, void * output_buffer,
                             long frame_count)
{
 EXPECT_TRUE(output_buffer);
 EXPECT_TRUE(!input_buffer);
 return frame_count;
}

TEST(cubeb, resampler_output_only_noop)
{
 cubeb_stream_params output_params;
 int target_rate;

 output_params.rate = 44100;
 output_params.channels = 1;
 output_params.format = CUBEB_SAMPLE_FLOAT32NE;
 target_rate = output_params.rate;

 cubeb_resampler * resampler = cubeb_resampler_create(
     (cubeb_stream *)nullptr, nullptr, &output_params, target_rate,
     test_output_only_noop_data_cb, nullptr, CUBEB_RESAMPLER_QUALITY_VOIP,
     CUBEB_RESAMPLER_RECLOCK_NONE);
 const long out_frames = 128;
 float out_buffer[out_frames];
 long got;

 got =
     cubeb_resampler_fill(resampler, nullptr, nullptr, out_buffer, out_frames);

 ASSERT_EQ(got, out_frames);

 cubeb_resampler_destroy(resampler);
}

long
test_drain_data_cb(cubeb_stream * /*stm*/, void * user_ptr,
                  const void * input_buffer, void * output_buffer,
                  long frame_count)
{
 EXPECT_TRUE(output_buffer);
 EXPECT_TRUE(!input_buffer);
 auto cb_count = static_cast<int *>(user_ptr);
 (*cb_count)++;
 return frame_count - 1;
}

TEST(cubeb, resampler_drain)
{
 cubeb_stream_params output_params;
 int target_rate;

 output_params.rate = 44100;
 output_params.channels = 1;
 output_params.format = CUBEB_SAMPLE_FLOAT32NE;
 target_rate = 48000;
 int cb_count = 0;

 cubeb_resampler * resampler = cubeb_resampler_create(
     (cubeb_stream *)nullptr, nullptr, &output_params, target_rate,
     test_drain_data_cb, &cb_count, CUBEB_RESAMPLER_QUALITY_VOIP,
     CUBEB_RESAMPLER_RECLOCK_NONE);

 const long out_frames = 128;
 float out_buffer[out_frames];
 long got;

 do {
   got = cubeb_resampler_fill(resampler, nullptr, nullptr, out_buffer,
                              out_frames);
 } while (got == out_frames);

 /* The callback should be called once but not again after returning <
  * frame_count. */
 ASSERT_EQ(cb_count, 1);

 cubeb_resampler_destroy(resampler);
}

// gtest does not support using ASSERT_EQ and friend in a function that returns
// a value.
void
check_output(const void * input_buffer, void * output_buffer, long frame_count)
{
 ASSERT_EQ(input_buffer, nullptr);
 ASSERT_EQ(frame_count, 256);
 ASSERT_TRUE(!!output_buffer);
}

long
cb_passthrough_resampler_output(cubeb_stream * /*stm*/, void * /*user_ptr*/,
                               const void * input_buffer, void * output_buffer,
                               long frame_count)
{
 check_output(input_buffer, output_buffer, frame_count);
 return frame_count;
}

TEST(cubeb, resampler_passthrough_output_only)
{
 // Test that the passthrough resampler works when there is only an output
 // stream.
 cubeb_stream_params output_params;

 const size_t output_channels = 2;
 output_params.channels = output_channels;
 output_params.rate = 44100;
 output_params.format = CUBEB_SAMPLE_FLOAT32NE;
 int target_rate = output_params.rate;

 cubeb_resampler * resampler = cubeb_resampler_create(
     (cubeb_stream *)nullptr, nullptr, &output_params, target_rate,
     cb_passthrough_resampler_output, nullptr, CUBEB_RESAMPLER_QUALITY_VOIP,
     CUBEB_RESAMPLER_RECLOCK_NONE);

 float output_buffer[output_channels * 256];

 long got;
 for (uint32_t i = 0; i < 30; i++) {
   got = cubeb_resampler_fill(resampler, nullptr, nullptr, output_buffer, 256);
   ASSERT_EQ(got, 256);
 }

 cubeb_resampler_destroy(resampler);
}

// gtest does not support using ASSERT_EQ and friend in a function that returns
// a value.
void
check_input(const void * input_buffer, void * output_buffer, long frame_count)
{
 ASSERT_EQ(output_buffer, nullptr);
 ASSERT_EQ(frame_count, 256);
 ASSERT_TRUE(!!input_buffer);
}

long
cb_passthrough_resampler_input(cubeb_stream * /*stm*/, void * /*user_ptr*/,
                              const void * input_buffer, void * output_buffer,
                              long frame_count)
{
 check_input(input_buffer, output_buffer, frame_count);
 return frame_count;
}

TEST(cubeb, resampler_passthrough_input_only)
{
 // Test that the passthrough resampler works when there is only an output
 // stream.
 cubeb_stream_params input_params;

 const size_t input_channels = 2;
 input_params.channels = input_channels;
 input_params.rate = 44100;
 input_params.format = CUBEB_SAMPLE_FLOAT32NE;
 int target_rate = input_params.rate;

 cubeb_resampler * resampler = cubeb_resampler_create(
     (cubeb_stream *)nullptr, &input_params, nullptr, target_rate,
     cb_passthrough_resampler_input, nullptr, CUBEB_RESAMPLER_QUALITY_VOIP,
     CUBEB_RESAMPLER_RECLOCK_NONE);

 float input_buffer[input_channels * 256];

 long got;
 for (uint32_t i = 0; i < 30; i++) {
   long int frames = 256;
   got = cubeb_resampler_fill(resampler, input_buffer, &frames, nullptr, 0);
   ASSERT_EQ(got, 256);
 }

 cubeb_resampler_destroy(resampler);
}

template <typename T>
long
seq(T * array, int stride, long start, long count)
{
 uint32_t output_idx = 0;
 for (int i = 0; i < count; i++) {
   for (int j = 0; j < stride; j++) {
     array[output_idx + j] = static_cast<T>(start + i);
   }
   output_idx += stride;
 }
 return start + count;
}

template <typename T>
void
is_seq(T * array, int stride, long count, long expected_start)
{
 uint32_t output_index = 0;
 for (long i = 0; i < count; i++) {
   for (int j = 0; j < stride; j++) {
     ASSERT_EQ(array[output_index + j], expected_start + i);
   }
   output_index += stride;
 }
}

template <typename T>
void
is_not_seq(T * array, int stride, long count, long expected_start)
{
 uint32_t output_index = 0;
 for (long i = 0; i < count; i++) {
   for (int j = 0; j < stride; j++) {
     ASSERT_NE(array[output_index + j], expected_start + i);
   }
   output_index += stride;
 }
}

struct closure {
 int input_channel_count;
};

// gtest does not support using ASSERT_EQ and friend in a function that returns
// a value.
template <typename T>
void
check_duplex(const T * input_buffer, T * output_buffer, long frame_count,
            int input_channel_count)
{
 ASSERT_EQ(frame_count, 256);
 // Silence scan-build warning.
 ASSERT_TRUE(!!output_buffer);
 assert(output_buffer);
 ASSERT_TRUE(!!input_buffer);
 assert(input_buffer);

 int output_index = 0;
 int input_index = 0;
 for (int i = 0; i < frame_count; i++) {
   // output is two channels, input one or two channels.
   if (input_channel_count == 1) {
     output_buffer[output_index] = output_buffer[output_index + 1] =
         input_buffer[i];
   } else if (input_channel_count == 2) {
     output_buffer[output_index] = input_buffer[input_index];
     output_buffer[output_index + 1] = input_buffer[input_index + 1];
   }
   output_index += 2;
   input_index += input_channel_count;
 }
}

long
cb_passthrough_resampler_duplex(cubeb_stream * /*stm*/, void * user_ptr,
                               const void * input_buffer, void * output_buffer,
                               long frame_count)
{
 closure * c = reinterpret_cast<closure *>(user_ptr);
 check_duplex<float>(static_cast<const float *>(input_buffer),
                     static_cast<float *>(output_buffer), frame_count,
                     c->input_channel_count);
 return frame_count;
}

TEST(cubeb, resampler_passthrough_duplex_callback_reordering)
{
 // Test that when pre-buffering on resampler creation, we can survive an input
 // callback being delayed.

 cubeb_stream_params input_params;
 cubeb_stream_params output_params;

 const int input_channels = 1;
 const int output_channels = 2;

 input_params.channels = input_channels;
 input_params.rate = 44100;
 input_params.format = CUBEB_SAMPLE_FLOAT32NE;

 output_params.channels = output_channels;
 output_params.rate = input_params.rate;
 output_params.format = CUBEB_SAMPLE_FLOAT32NE;

 int target_rate = input_params.rate;

 closure c;
 c.input_channel_count = input_channels;

 cubeb_resampler * resampler = cubeb_resampler_create(
     (cubeb_stream *)nullptr, &input_params, &output_params, target_rate,
     cb_passthrough_resampler_duplex, &c, CUBEB_RESAMPLER_QUALITY_VOIP,
     CUBEB_RESAMPLER_RECLOCK_NONE);

 const long BUF_BASE_SIZE = 256;
 float input_buffer_prebuffer[input_channels * BUF_BASE_SIZE * 2];
 float input_buffer_glitch[input_channels * BUF_BASE_SIZE * 2];
 float input_buffer_normal[input_channels * BUF_BASE_SIZE];
 float output_buffer[output_channels * BUF_BASE_SIZE];

 long seq_idx = 0;
 long output_seq_idx = 0;

 long prebuffer_frames =
     ARRAY_LENGTH(input_buffer_prebuffer) / input_params.channels;
 seq_idx =
     seq(input_buffer_prebuffer, input_channels, seq_idx, prebuffer_frames);

 long got =
     cubeb_resampler_fill(resampler, input_buffer_prebuffer, &prebuffer_frames,
                          output_buffer, BUF_BASE_SIZE);

 output_seq_idx += BUF_BASE_SIZE;

 // prebuffer_frames will hold the frames used by the resampler.
 ASSERT_EQ(prebuffer_frames, BUF_BASE_SIZE);
 ASSERT_EQ(got, BUF_BASE_SIZE);

 for (uint32_t i = 0; i < 300; i++) {
   long int frames = BUF_BASE_SIZE;
   // Simulate that sometimes, we don't have the input callback on time
   if (i != 0 && (i % 100) == 0) {
     long zero = 0;
     got =
         cubeb_resampler_fill(resampler, input_buffer_normal /* unused here */,
                              &zero, output_buffer, BUF_BASE_SIZE);
     is_seq(output_buffer, 2, BUF_BASE_SIZE, output_seq_idx);
     output_seq_idx += BUF_BASE_SIZE;
   } else if (i != 0 && (i % 100) == 1) {
     // if this is the case, the on the next iteration, we'll have twice the
     // amount of input frames
     seq_idx =
         seq(input_buffer_glitch, input_channels, seq_idx, BUF_BASE_SIZE * 2);
     frames = 2 * BUF_BASE_SIZE;
     got = cubeb_resampler_fill(resampler, input_buffer_glitch, &frames,
                                output_buffer, BUF_BASE_SIZE);
     is_seq(output_buffer, 2, BUF_BASE_SIZE, output_seq_idx);
     output_seq_idx += BUF_BASE_SIZE;
   } else {
     // normal case
     seq_idx =
         seq(input_buffer_normal, input_channels, seq_idx, BUF_BASE_SIZE);
     long normal_input_frame_count = 256;
     got = cubeb_resampler_fill(resampler, input_buffer_normal,
                                &normal_input_frame_count, output_buffer,
                                BUF_BASE_SIZE);
     is_seq(output_buffer, 2, BUF_BASE_SIZE, output_seq_idx);
     output_seq_idx += BUF_BASE_SIZE;
   }
   ASSERT_EQ(got, BUF_BASE_SIZE);
 }

 cubeb_resampler_destroy(resampler);
}

// Artificially simulate output thread underruns,
// by building up artificial delay in the input.
// Check that the frame drop logic kicks in.
TEST(cubeb, resampler_drift_drop_data)
{
 for (uint32_t input_channels = 1; input_channels < 3; input_channels++) {
   cubeb_stream_params input_params;
   cubeb_stream_params output_params;

   const int output_channels = 2;
   const int sample_rate = 44100;

   input_params.channels = input_channels;
   input_params.rate = sample_rate;
   input_params.format = CUBEB_SAMPLE_FLOAT32NE;

   output_params.channels = output_channels;
   output_params.rate = sample_rate;
   output_params.format = CUBEB_SAMPLE_FLOAT32NE;

   int target_rate = input_params.rate;

   closure c;
   c.input_channel_count = input_channels;

   cubeb_resampler * resampler = cubeb_resampler_create(
       (cubeb_stream *)nullptr, &input_params, &output_params, target_rate,
       cb_passthrough_resampler_duplex, &c, CUBEB_RESAMPLER_QUALITY_VOIP,
       CUBEB_RESAMPLER_RECLOCK_NONE);

   const long BUF_BASE_SIZE = 256;

   // The factor by which the deadline is missed. This is intentionally
   // kind of large to trigger the frame drop quickly. In real life, multiple
   // smaller under-runs would accumulate.
   const long UNDERRUN_FACTOR = 10;
   // Number buffer used for pre-buffering, that some backends do.
   const long PREBUFFER_FACTOR = 2;

   std::vector<float> input_buffer_prebuffer(input_channels * BUF_BASE_SIZE *
                                             PREBUFFER_FACTOR);
   std::vector<float> input_buffer_glitch(input_channels * BUF_BASE_SIZE *
                                          UNDERRUN_FACTOR);
   std::vector<float> input_buffer_normal(input_channels * BUF_BASE_SIZE);
   std::vector<float> output_buffer(output_channels * BUF_BASE_SIZE);

   long seq_idx = 0;
   long output_seq_idx = 0;

   long prebuffer_frames =
       input_buffer_prebuffer.size() / input_params.channels;
   seq_idx = seq(input_buffer_prebuffer.data(), input_channels, seq_idx,
                 prebuffer_frames);

   long got = cubeb_resampler_fill(resampler, input_buffer_prebuffer.data(),
                                   &prebuffer_frames, output_buffer.data(),
                                   BUF_BASE_SIZE);

   output_seq_idx += BUF_BASE_SIZE;

   // prebuffer_frames will hold the frames used by the resampler.
   ASSERT_EQ(prebuffer_frames, BUF_BASE_SIZE);
   ASSERT_EQ(got, BUF_BASE_SIZE);

   for (uint32_t i = 0; i < 300; i++) {
     long int frames = BUF_BASE_SIZE;
     if (i != 0 && (i % 100) == 1) {
       // Once in a while, the output thread misses its deadline.
       // The input thread still produces data, so it ends up accumulating.
       // Simulate this by providing a much bigger input buffer. Check that the
       // sequence is now unaligned, meaning we've dropped data to keep
       // everything in sync.
       seq_idx = seq(input_buffer_glitch.data(), input_channels, seq_idx,
                     BUF_BASE_SIZE * UNDERRUN_FACTOR);
       frames = BUF_BASE_SIZE * UNDERRUN_FACTOR;
       got =
           cubeb_resampler_fill(resampler, input_buffer_glitch.data(), &frames,
                                output_buffer.data(), BUF_BASE_SIZE);
       is_seq(output_buffer.data(), 2, BUF_BASE_SIZE, output_seq_idx);
       output_seq_idx += BUF_BASE_SIZE;
     } else if (i != 0 && (i % 100) == 2) {
       // On the next iteration, the sequence should be broken
       seq_idx = seq(input_buffer_normal.data(), input_channels, seq_idx,
                     BUF_BASE_SIZE);
       long normal_input_frame_count = 256;
       got = cubeb_resampler_fill(resampler, input_buffer_normal.data(),
                                  &normal_input_frame_count,
                                  output_buffer.data(), BUF_BASE_SIZE);
       is_not_seq(output_buffer.data(), output_channels, BUF_BASE_SIZE,
                  output_seq_idx);
       // Reclock so that we can use is_seq again.
       output_seq_idx = output_buffer[BUF_BASE_SIZE * output_channels - 1] + 1;
     } else {
       // normal case
       seq_idx = seq(input_buffer_normal.data(), input_channels, seq_idx,
                     BUF_BASE_SIZE);
       long normal_input_frame_count = 256;
       got = cubeb_resampler_fill(resampler, input_buffer_normal.data(),
                                  &normal_input_frame_count,
                                  output_buffer.data(), BUF_BASE_SIZE);
       is_seq(output_buffer.data(), output_channels, BUF_BASE_SIZE,
              output_seq_idx);
       output_seq_idx += BUF_BASE_SIZE;
     }
     ASSERT_EQ(got, BUF_BASE_SIZE);
   }

   cubeb_resampler_destroy(resampler);
 }
}

static long
passthrough_resampler_fill_eq_input(cubeb_stream * stream, void * user_ptr,
                                   void const * input_buffer,
                                   void * output_buffer, long nframes)
{
 // gtest does not support using ASSERT_EQ and friends in a
 // function that returns a value.
 [nframes, input_buffer]() {
   ASSERT_EQ(nframes, 32);
   const float * input = static_cast<const float *>(input_buffer);
   for (int i = 0; i < 64; ++i) {
     ASSERT_FLOAT_EQ(input[i], 0.01 * i);
   }
 }();
 return nframes;
}

TEST(cubeb, passthrough_resampler_fill_eq_input)
{
 uint32_t channels = 2;
 uint32_t sample_rate = 44100;
 passthrough_resampler<float> resampler =
     passthrough_resampler<float>(nullptr, passthrough_resampler_fill_eq_input,
                                  nullptr, channels, sample_rate);

 long input_frame_count = 32;
 long output_frame_count = 32;
 float input[64] = {};
 float output[64] = {};
 for (uint32_t i = 0; i < input_frame_count * channels; ++i) {
   input[i] = 0.01 * i;
 }
 long got =
     resampler.fill(input, &input_frame_count, output, output_frame_count);
 ASSERT_EQ(got, output_frame_count);
 // Input frames used must be equal to output frames.
 ASSERT_EQ(input_frame_count, output_frame_count);
}

static long
passthrough_resampler_fill_short_input(cubeb_stream * stream, void * user_ptr,
                                      void const * input_buffer,
                                      void * output_buffer, long nframes)
{
 // gtest does not support using ASSERT_EQ and friends in a
 // function that returns a value.
 [nframes, input_buffer]() {
   ASSERT_EQ(nframes, 32);
   const float * input = static_cast<const float *>(input_buffer);
   // First part contains the input
   for (int i = 0; i < 32; ++i) {
     ASSERT_FLOAT_EQ(input[i], 0.01 * i);
   }
   // missing part contains silence
   for (int i = 32; i < 64; ++i) {
     ASSERT_FLOAT_EQ(input[i], 0.0);
   }
 }();
 return nframes;
}

TEST(cubeb, passthrough_resampler_fill_short_input)
{
 uint32_t channels = 2;
 uint32_t sample_rate = 44100;
 passthrough_resampler<float> resampler = passthrough_resampler<float>(
     nullptr, passthrough_resampler_fill_short_input, nullptr, channels,
     sample_rate);

 long input_frame_count = 16;
 long output_frame_count = 32;
 float input[64] = {};
 float output[64] = {};
 for (uint32_t i = 0; i < input_frame_count * channels; ++i) {
   input[i] = 0.01 * i;
 }
 long got =
     resampler.fill(input, &input_frame_count, output, output_frame_count);
 ASSERT_EQ(got, output_frame_count);
 // Input frames used are less than the output frames due to glitch.
 ASSERT_EQ(input_frame_count, output_frame_count - 16);
}

static long
passthrough_resampler_fill_input_left(cubeb_stream * stream, void * user_ptr,
                                     void const * input_buffer,
                                     void * output_buffer, long nframes)
{
 // gtest does not support using ASSERT_EQ and friends in a
 // function that returns a value.
 int iteration = *static_cast<int *>(user_ptr);
 if (iteration == 1) {
   [nframes, input_buffer]() {
     ASSERT_EQ(nframes, 32);
     const float * input = static_cast<const float *>(input_buffer);
     for (int i = 0; i < 64; ++i) {
       ASSERT_FLOAT_EQ(input[i], 0.01 * i);
     }
   }();
 } else if (iteration == 2) {
   [nframes, input_buffer]() {
     ASSERT_EQ(nframes, 32);
     const float * input = static_cast<const float *>(input_buffer);
     for (int i = 0; i < 32; ++i) {
       // First part contains the reamaining input samples from previous
       // iteration (since they were more).
       ASSERT_FLOAT_EQ(input[i], 0.01 * (i + 64));
       // next part contains the new buffer
       ASSERT_FLOAT_EQ(input[i + 32], 0.01 * i);
     }
   }();
 } else if (iteration == 3) {
   [nframes, input_buffer]() {
     ASSERT_EQ(nframes, 32);
     const float * input = static_cast<const float *>(input_buffer);
     for (int i = 0; i < 32; ++i) {
       // First part (16 frames) contains the reamaining input samples
       // from previous iteration (since they were more).
       ASSERT_FLOAT_EQ(input[i], 0.01 * (i + 32));
     }
     for (int i = 0; i < 16; ++i) {
       // next part (8 frames) contains the new input buffer.
       ASSERT_FLOAT_EQ(input[i + 32], 0.01 * i);
       // last part (8 frames) contains silence.
       ASSERT_FLOAT_EQ(input[i + 32 + 16], 0.0);
     }
   }();
 }
 return nframes;
}

TEST(cubeb, passthrough_resampler_fill_input_left)
{
 const uint32_t channels = 2;
 const uint32_t sample_rate = 44100;
 int iteration = 0;
 passthrough_resampler<float> resampler = passthrough_resampler<float>(
     nullptr, passthrough_resampler_fill_input_left, &iteration, channels,
     sample_rate);

 long input_frame_count = 48; // 32 + 16
 const long output_frame_count = 32;
 float input[96] = {};
 float output[64] = {};
 for (uint32_t i = 0; i < input_frame_count * channels; ++i) {
   input[i] = 0.01 * i;
 }

 // 1st iteration, add the extra input.
 iteration = 1;
 long got =
     resampler.fill(input, &input_frame_count, output, output_frame_count);
 ASSERT_EQ(got, output_frame_count);
 // Input frames used must be equal to output frames.
 ASSERT_EQ(input_frame_count, output_frame_count);

 // 2st iteration, use the extra input from previous iteration,
 // 16 frames are remaining in the input buffer.
 input_frame_count = 32; // we need 16 input frames but we get more;
 iteration = 2;
 got = resampler.fill(input, &input_frame_count, output, output_frame_count);
 ASSERT_EQ(got, output_frame_count);
 // Input frames used must be equal to output frames.
 ASSERT_EQ(input_frame_count, output_frame_count);

 // 3rd iteration, use the extra input from previous iteration.
 // 16 frames are remaining in the input buffer.
 input_frame_count = 16 - 8; // We need 16 more input frames but we only get 8.
 iteration = 3;
 got = resampler.fill(input, &input_frame_count, output, output_frame_count);
 ASSERT_EQ(got, output_frame_count);
 // Input frames used are less than the output frames due to glitch.
 ASSERT_EQ(input_frame_count, output_frame_count - 8);
}

TEST(cubeb, individual_methods)
{
 const uint32_t channels = 2;
 const uint32_t sample_rate = 44100;
 const uint32_t frames = 256;

 delay_line<float> dl(10, channels, sample_rate);
 uint32_t frames_needed1 = dl.input_needed_for_output(0);
 ASSERT_EQ(frames_needed1, 0u);

 cubeb_resampler_speex_one_way<float> one_way(
     channels, sample_rate, sample_rate, CUBEB_RESAMPLER_QUALITY_DEFAULT);
 float buffer[channels * frames] = {0.0};
 // Add all frames in the resampler's internal buffer.
 one_way.input(buffer, frames);
 // Ask for less than the existing frames, this would create a uint overlflow
 // without the fix.
 uint32_t frames_needed2 = one_way.input_needed_for_output(0);
 ASSERT_EQ(frames_needed2, 0u);
}

struct sine_wave_state {
 float frequency;
 int sample_rate;
 size_t count = 0;
 sine_wave_state(float freq, int rate) : frequency(freq), sample_rate(rate) {}
};

long
data_cb(cubeb_stream * stream, void * user_ptr, void const * input_buffer,
       void * output_buffer, long nframes)
{
 sine_wave_state * state = static_cast<sine_wave_state *>(user_ptr);
 float * out = static_cast<float *>(output_buffer);
 double phase_increment = 2.0f * M_PI * state->frequency / state->sample_rate;

 for (int i = 0; i < nframes; i++) {
   float sample = sin(phase_increment * state->count);
   state->count++;
   out[i] = sample * 0.8;
 }
 return nframes;
}

// This implements 4.6.2 from "Standard for Digitizing Waveform Recorders"
// (in particular Annex A), then returns the estimated amplitude, phase, and the
// sum of squared error relative to a sine wave sampled at `sample_rate` and of
// frequency `frequency`. This is also described in "Numerical methods for
// engineers" chapter 19.1, and explained at
// https://www.youtube.com/watch?v=afQszl_OwKo and videos of the same series.
// In practice here we're sending a perfect 1khz sine wave into a good
// resampler, and despite the resampling ratio being quite extreme sometimes,
// we're expecting a very good fit.
float
fit_sine(const std::vector<float> & signal, float sample_rate, float frequency,
        float & out_amplitude, float & out_phase)
{
 // The formulation below is exact for samples spanning an integer number of
 // periods. It can be important for `signal` to be trimmed to an integer
 // number of periods if it doesn't contain a lot of periods.
 double phase_incr = 2.0 * M_PI * frequency / sample_rate;

 double sum_cos = 0.0;
 double sum_sin = 0.0;
 for (size_t i = 0; i < signal.size(); ++i) {
   double c = std::cos(phase_incr * static_cast<double>(i));
   double s = std::sin(phase_incr * static_cast<double>(i));
   sum_cos += signal[i] * c;
   sum_sin += signal[i] * s;
 }

 double amplitude = 2.0f * std::sqrt(sum_cos * sum_cos + sum_sin * sum_sin) /
                    static_cast<double>(signal.size());
 double phi = std::atan2(sum_cos, sum_sin);

 out_amplitude = amplitude;
 out_phase = phi;

 // Compute sum of squared errors relative to the fitted sine wave
 double sse = 0.0;
 for (size_t i = 0; i < signal.size(); ++i) {
   // Use known amplitude here instead instead of the from the fitted function.
   double fit = 0.8 * std::sin(phase_incr * i + phi);
   double diff = signal[i] - fit;
   sse += diff * diff;
 }

 return sse;
}

// Finds the offset of the start of an input_freq sine wave sampled at
// target_rate in data. Remove the leading silence from data.
size_t
find_sine_start(const std::vector<float> & data, float input_freq,
               float target_rate)
{
 const size_t POINTS = 10;
 size_t skipped = 0;

 while (skipped + POINTS < data.size()) {
   double phase = 0;
   double phase_increment = 2.0f * M_PI * input_freq / target_rate;
   bool fits_sine = true;

   for (size_t i = 0; i < POINTS; i++) {
     float expected = sin(phase) * 0.8;
     float actual = data[skipped + i];
     if (fabs(expected - actual) > 0.1) {
       // doesn't fit a sine, skip to next start point
       fits_sine = false;
       break;
     }
     phase += phase_increment;
     if (phase > 2.0f * M_PI) {
       phase -= 2.0f * M_PI;
     }
   }

   if (!fits_sine) {
     skipped++;
     continue;
   }

   // Found the start of the sine wave
   size_t sine_start = skipped;
   return sine_start;
 }

 return skipped;
}

// This class tracks the monotonicity of a certain value, and reports if it
// increases too much monotonically.
struct monotonic_state {
 explicit monotonic_state(const char * what, int source_rate, int target_rate,
                          int block_size)
     : what(what), source_rate(source_rate), target_rate(target_rate),
       block_size(block_size)
 {
 }
 ~monotonic_state()
 {
   float ratio =
       static_cast<float>(source_rate) / static_cast<float>(target_rate);
   // Only report if there has been a meaningful increase in buffering. Do
   // not warn if the buffering was constant and small.
   if (monotonic && max_value && max_value != max_step) {
     printf("%s is monotonically increasing, max: %zu, max_step: %zu, "
            "in: %dHz, out: "
            "%dHz, block_size: %d, ratio: %lf\n",
            what, max_value, max_step, source_rate, target_rate, block_size,
            ratio);
   }
   // Arbitrary limit: if more than this number of frames has been buffered,
   // print a message.
   constexpr int BUFFER_SIZE_THRESHOLD = 20;
   if (max_value > BUFFER_SIZE_THRESHOLD) {
     printf("%s, unexpected large max buffering value, max: %zu, max_step: "
            "%zu, in: %dHz, out: %dHz, block_size: %d, ratio: %lf\n",
            what, max_value, max_step, source_rate, target_rate, block_size,
            ratio);
   }
 }
 void set_new_value(size_t new_value)
 {
   if (new_value < value) {
     monotonic = false;
   } else {
     max_step = std::max(max_step, new_value - value);
   }
   value = new_value;
   max_value = std::max(value, max_value);
 }
 // Textual representation of this measurement
 const char * what;
 // Resampler parameters for this test case
 int source_rate = 0;
 int target_rate = 0;
 int block_size = 0;
 // Current buffering value
 size_t value = 0;
 // Max buffering value increment
 size_t max_step = 0;
 // Max buffering value observerd
 size_t max_value = 0;
 // Whether the value has only increased or not
 bool monotonic = true;
};

// Setting this to 1 dumps a bunch of wave file to the local directory for
// manual inspection of the resampled output
constexpr int DUMP_OUTPUT = 0;

// Source and target sample-rates in Hz, typical values.
const int rates[] = {16000, 32000, 44100, 48000, 96000, 192000, 384000};
// Block size in frames, except the first element, that is in millisecond
// Power of two are typical on Windows WASAPI IAudioClient3, macOS,
// Linux Pipewire and Jack. 10ms is typical on Windows IAudioClient and
// IAudioClient2. 96, 192 are not uncommon on some Android devices.
constexpr int WASAPI_MS_BLOCK = 10;
const int block_sizes[] = {WASAPI_MS_BLOCK, 96, 128, 192, 256, 512, 1024, 2048};
// Enough iterations to catch rounding/drift issues, but not too many to avoid
// having a test that is too long to run.
constexpr int ITERATION_COUNT = 1000;
// 1 kHz input sine wave
const float input_freq = 1000.0f;

struct ThreadPool {
 std::vector<std::thread> workers;
 std::queue<std::function<void()>> tasks;
 std::mutex queue_mutex;
 std::condition_variable condition;
 bool stop;

 ThreadPool(size_t threads) : stop(false)
 {
   for (size_t i = 0; i < threads; ++i) {
     workers.emplace_back([this] {
       while (true) {
         std::function<void()> task;
         {
           std::unique_lock<std::mutex> lock(queue_mutex);
           condition.wait(lock, [this] { return stop || !tasks.empty(); });
           if (stop && tasks.empty())
             return;
           task = std::move(tasks.front());
           tasks.pop();
         }
         task();
       }
     });
   }
 }

 void enqueue(std::function<void()> task)
 {
   {
     std::unique_lock<std::mutex> lock(queue_mutex);
     tasks.push(std::move(task));
   }
   condition.notify_one();
 }

 ~ThreadPool()
 {
   {
     std::unique_lock<std::mutex> lock(queue_mutex);
     stop = true;
   }
   condition.notify_all();
   for (std::thread & worker : workers) {
     worker.join();
   }
 }
};

static void
run_test(int source_rate, int target_rate, int block_size)
{
 int effective_block_size = block_size;
 // special case: Windows/WASAPI works in blocks of 10ms regardless of
 // the rate.
 if (effective_block_size == WASAPI_MS_BLOCK) {
   effective_block_size = target_rate / 100; // 10ms
 }
 sine_wave_state state(input_freq, source_rate);
 cubeb_stream_params out_params = {};
 out_params.channels = 1;
 out_params.rate = target_rate;
 out_params.format = CUBEB_SAMPLE_FLOAT32NE;

 cubeb_audio_dump_session_t session = nullptr;
 cubeb_audio_dump_stream_t dump_stream = nullptr;
 if constexpr (DUMP_OUTPUT) {
   cubeb_audio_dump_init(&session);
   char buf[256];
   snprintf(buf, 256, "test-%dHz-to-%dhz-%d-block.wav", source_rate,
            target_rate, effective_block_size);
   cubeb_audio_dump_stream_init(session, &dump_stream, out_params, buf);
   cubeb_audio_dump_start(session);
 }
 cubeb_resampler * resampler = cubeb_resampler_create(
     nullptr, nullptr, &out_params, source_rate, data_cb, &state,
     CUBEB_RESAMPLER_QUALITY_DEFAULT, CUBEB_RESAMPLER_RECLOCK_NONE);
 ASSERT_NE(resampler, nullptr);

 std::vector<float> data(effective_block_size * out_params.channels);
 int i = ITERATION_COUNT;
 // For now this only tests the output side (out_... measurements).
 //  We could expect the resampler to be symmetrical, but we could
 //  test both sides at once.
 // - ..._in is the input buffer of the resampler, containing
 // unresampled  frames
 // - ..._out is the output buffer, containing resampled frames.
 monotonic_state in_in_max("in_in", source_rate, target_rate,
                           effective_block_size);
 monotonic_state in_out_max("in_out", source_rate, target_rate,
                            effective_block_size);
 monotonic_state out_in_max("out_in", source_rate, target_rate,
                            effective_block_size);
 monotonic_state out_out_max("out_out", source_rate, target_rate,
                             effective_block_size);

 std::vector<float> resampled;
 resampled.reserve(ITERATION_COUNT * effective_block_size *
                   out_params.channels);
 while (i--) {
   int64_t got = cubeb_resampler_fill(resampler, nullptr, nullptr, data.data(),
                                      effective_block_size);
   ASSERT_EQ(got, effective_block_size);
   cubeb_resampler_stats stats = cubeb_resampler_stats_get(resampler);

   resampled.insert(resampled.end(), data.begin(), data.end());

   in_in_max.set_new_value(stats.input_input_buffer_size);
   in_out_max.set_new_value(stats.input_output_buffer_size);
   out_in_max.set_new_value(stats.output_input_buffer_size);
   out_out_max.set_new_value(stats.output_output_buffer_size);
 }

 cubeb_resampler_destroy(resampler);

 // Example of an error, off by one every block or so, resulting in a
 // silent sample. This is enough to make all the tests fail.
 //
 // for (uint32_t i = 0; i < resampled.size(); i++) {
 //   if (!(i % (effective_block_size))) {
 //     resampled[i] = 0.0;
 //   }
 // }

 // This roughly finds the start of the sine wave and strips it from
 // data.
 size_t skipped = 0;
 skipped = find_sine_start(resampled, input_freq, target_rate);

 resampled.erase(resampled.begin(), resampled.begin() + skipped);

 if constexpr (DUMP_OUTPUT) {
   cubeb_audio_dump_write(dump_stream, resampled.data(), resampled.size());
 }

 float amplitude = 0;
 float phase = 0;

 // Fit our resampled sine wave, get an MSE value
 double sse = fit_sine(resampled, target_rate, input_freq, amplitude, phase);
 double mse = sse / resampled.size();

 // Code to print JSON to plot externally
 // printf("\t[%d,%d,%d,%.10e,%lf,%lf],\n", source_rate, target_rate,
 //        effective_block_size, mse, amplitude, phase);

 // Value found after running the tests on Linux x64
 ASSERT_LT(mse, 3.22e-07);

 if constexpr (DUMP_OUTPUT) {
   cubeb_audio_dump_stop(session);
   cubeb_audio_dump_stream_shutdown(session, dump_stream);
   cubeb_audio_dump_shutdown(session);
 }
}

// This tests checks three things:
// - Whenever resampling from a source rate to a target rate with a certain
//  block size, the correct number of frames is provided back from the
//  resampler, to the backend.
// - While resampling, internal buffers are kept under control and aren't
// growing unbounded.
// - The output signal is a 1khz sine (as is the input)
TEST(cubeb, resampler_typical_uses)
{
 cubeb * ctx;
 common_init(&ctx, "Cubeb resampler test");

 size_t concurrency = std::max(1u, std::thread::hardware_concurrency());
 std::condition_variable cv;
 std::mutex mutex;
 size_t task_count = 0;
 ThreadPool pool(concurrency);

 for (int source_rate : rates) {
   for (int target_rate : rates) {
     for (int block_size : block_sizes) {
       {
         std::unique_lock<std::mutex> lock(mutex);
         ++task_count;
       }
       pool.enqueue([&, source_rate, target_rate, block_size] {
         run_test(source_rate, target_rate, block_size);
         {
           std::unique_lock<std::mutex> lock(mutex);
           --task_count;
         }
         cv.notify_one();
       });
     }
   }
 }

 std::unique_lock<std::mutex> lock(mutex);
 cv.wait(lock, [&] { return task_count == 0; });
 cubeb_destroy(ctx);
}
#undef NOMINMAX
#undef DUMP_ARRAYS