webrtc/audio/remix_resample_unittest.cc

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "audio/remix_resample.h"

#include <cmath>

#include "common_audio/resampler/include/push_resampler.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"

namespace webrtc {
namespace voe {
namespace {

int GetFrameSize(int sample_rate_hz) {
  return sample_rate_hz / 100;
}

class UtilityTest : public ::testing::Test {
 protected:
  UtilityTest() {
    src_frame_.sample_rate_hz_ = 16000;
    src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100;
    src_frame_.num_channels_ = 1;
    dst_frame_.CopyFrom(src_frame_);
    golden_frame_.CopyFrom(src_frame_);
  }

  void RunResampleTest(int src_channels,
                       int src_sample_rate_hz,
                       int dst_channels,
                       int dst_sample_rate_hz);

  PushResampler<int16_t> resampler_;
  AudioFrame src_frame_;
  AudioFrame dst_frame_;
  AudioFrame golden_frame_;
};

// Sets the signal value to increase by `data` with every sample. Floats are
// used so non-integer values result in rounding error, but not an accumulating
// error.
void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) {
  frame->Mute();
  frame->num_channels_ = 1;
  frame->sample_rate_hz_ = sample_rate_hz;
  frame->samples_per_channel_ = GetFrameSize(sample_rate_hz);
  int16_t* frame_data = frame->mutable_data();
  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
    frame_data[i] = static_cast<int16_t>(data * i);
  }
}

// Keep the existing sample rate.
void SetMonoFrame(float data, AudioFrame* frame) {
  SetMonoFrame(data, frame->sample_rate_hz_, frame);
}

// Sets the signal value to increase by `left` and `right` with every sample in
// each channel respectively.
void SetStereoFrame(float left,
                    float right,
                    int sample_rate_hz,
                    AudioFrame* frame) {
  frame->Mute();
  frame->num_channels_ = 2;
  frame->sample_rate_hz_ = sample_rate_hz;
  frame->samples_per_channel_ = GetFrameSize(sample_rate_hz);
  int16_t* frame_data = frame->mutable_data();
  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
    frame_data[i * 2] = static_cast<int16_t>(left * i);
    frame_data[i * 2 + 1] = static_cast<int16_t>(right * i);
  }
}

// Keep the existing sample rate.
void SetStereoFrame(float left, float right, AudioFrame* frame) {
  SetStereoFrame(left, right, frame->sample_rate_hz_, frame);
}

// Sets the signal value to increase by `ch1`, `ch2`, `ch3`, `ch4` with every
// sample in each channel respectively.
void SetQuadFrame(float ch1,
                  float ch2,
                  float ch3,
                  float ch4,
                  int sample_rate_hz,
                  AudioFrame* frame) {
  frame->Mute();
  frame->num_channels_ = 4;
  frame->sample_rate_hz_ = sample_rate_hz;
  frame->samples_per_channel_ = GetFrameSize(sample_rate_hz);
  int16_t* frame_data = frame->mutable_data();
  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
    frame_data[i * 4] = static_cast<int16_t>(ch1 * i);
    frame_data[i * 4 + 1] = static_cast<int16_t>(ch2 * i);
    frame_data[i * 4 + 2] = static_cast<int16_t>(ch3 * i);
    frame_data[i * 4 + 3] = static_cast<int16_t>(ch4 * i);
  }
}

void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
  EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_);
  EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_);
  EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
}

// Computes the best SNR based on the error between `ref_frame` and
// `test_frame`. It allows for up to a `max_delay` in samples between the
// signals to compensate for the resampling delay.
float ComputeSNR(const AudioFrame& ref_frame,
                 const AudioFrame& test_frame,
                 size_t max_delay) {
  VerifyParams(ref_frame, test_frame);
  float best_snr = 0;
  size_t best_delay = 0;
  for (size_t delay = 0; delay <= max_delay; delay++) {
    float mse = 0;
    float variance = 0;
    const int16_t* ref_frame_data = ref_frame.data();
    const int16_t* test_frame_data = test_frame.data();
    for (size_t i = 0;
         i < ref_frame.samples_per_channel_ * ref_frame.num_channels_ - delay;
         i++) {
      int error = ref_frame_data[i] - test_frame_data[i + delay];
      mse += error * error;
      variance += ref_frame_data[i] * ref_frame_data[i];
    }
    float snr = 100;  // We assign 100 dB to the zero-error case.
    if (mse > 0)
      snr = 10 * std::log10(variance / mse);
    if (snr > best_snr) {
      best_snr = snr;
      best_delay = delay;
    }
  }
  printf("SNR=%.1f dB at delay=%zu\n", best_snr, best_delay);
  return best_snr;
}

void VerifyFramesAreEqual(const AudioFrame& ref_frame,
                          const AudioFrame& test_frame) {
  VerifyParams(ref_frame, test_frame);
  const int16_t* ref_frame_data = ref_frame.data();
  const int16_t* test_frame_data = test_frame.data();
  for (size_t i = 0;
       i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
    EXPECT_EQ(ref_frame_data[i], test_frame_data[i]);
  }
}

void UtilityTest::RunResampleTest(int src_channels,
                                  int src_sample_rate_hz,
                                  int dst_channels,
                                  int dst_sample_rate_hz) {
  PushResampler<int16_t> resampler;  // Create a new one with every test.
  const int16_t kSrcCh1 = 30;  // Shouldn't overflow for any used sample rate.
  const int16_t kSrcCh2 = 15;
  const int16_t kSrcCh3 = 22;
  const int16_t kSrcCh4 = 8;
  const float resampling_factor =
      (1.0 * src_sample_rate_hz) / dst_sample_rate_hz;
  const float dst_ch1 = resampling_factor * kSrcCh1;
  const float dst_ch2 = resampling_factor * kSrcCh2;
  const float dst_ch3 = resampling_factor * kSrcCh3;
  const float dst_ch4 = resampling_factor * kSrcCh4;
  const float dst_stereo_to_mono = (dst_ch1 + dst_ch2) / 2;
  const float dst_quad_to_mono = (dst_ch1 + dst_ch2 + dst_ch3 + dst_ch4) / 4;
  const float dst_quad_to_stereo_ch1 = (dst_ch1 + dst_ch2) / 2;
  const float dst_quad_to_stereo_ch2 = (dst_ch3 + dst_ch4) / 2;
  if (src_channels == 1)
    SetMonoFrame(kSrcCh1, src_sample_rate_hz, &src_frame_);
  else if (src_channels == 2)
    SetStereoFrame(kSrcCh1, kSrcCh2, src_sample_rate_hz, &src_frame_);
  else
    SetQuadFrame(kSrcCh1, kSrcCh2, kSrcCh3, kSrcCh4, src_sample_rate_hz,
                 &src_frame_);

  if (dst_channels == 1) {
    SetMonoFrame(0, dst_sample_rate_hz, &dst_frame_);
    if (src_channels == 1)
      SetMonoFrame(dst_ch1, dst_sample_rate_hz, &golden_frame_);
    else if (src_channels == 2)
      SetMonoFrame(dst_stereo_to_mono, dst_sample_rate_hz, &golden_frame_);
    else
      SetMonoFrame(dst_quad_to_mono, dst_sample_rate_hz, &golden_frame_);
  } else {
    SetStereoFrame(0, 0, dst_sample_rate_hz, &dst_frame_);
    if (src_channels == 1)
      SetStereoFrame(dst_ch1, dst_ch1, dst_sample_rate_hz, &golden_frame_);
    else if (src_channels == 2)
      SetStereoFrame(dst_ch1, dst_ch2, dst_sample_rate_hz, &golden_frame_);
    else
      SetStereoFrame(dst_quad_to_stereo_ch1, dst_quad_to_stereo_ch2,
                     dst_sample_rate_hz, &golden_frame_);
  }

  // The sinc resampler has a known delay, which we compute here. Multiplying by
  // two gives us a crude maximum for any resampling, as the old resampler
  // typically (but not always) has lower delay.
  static const size_t kInputKernelDelaySamples = 16;
  const size_t max_delay = static_cast<size_t>(
      static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz *
      kInputKernelDelaySamples * dst_channels * 2);
  printf("(%d, %d Hz) -> (%d, %d Hz) ",  // SNR reported on the same line later.
         src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
  RemixAndResample(src_frame_, &resampler, &dst_frame_);

  if (src_sample_rate_hz == 96000 && dst_sample_rate_hz <= 11025) {
    // The sinc resampler gives poor SNR at this extreme conversion, but we
    // expect to see this rarely in practice.
    EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 14.0f);
  } else {
    EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 46.0f);
  }
}

TEST_F(UtilityTest, RemixAndResampleCopyFrameSucceeds) {
  // Stereo -> stereo.
  SetStereoFrame(10, 10, &src_frame_);
  SetStereoFrame(0, 0, &dst_frame_);
  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
  VerifyFramesAreEqual(src_frame_, dst_frame_);

  // Mono -> mono.
  SetMonoFrame(20, &src_frame_);
  SetMonoFrame(0, &dst_frame_);
  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
  VerifyFramesAreEqual(src_frame_, dst_frame_);
}

TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) {
  // Stereo -> mono.
  SetStereoFrame(0, 0, &dst_frame_);
  SetMonoFrame(10, &src_frame_);
  SetStereoFrame(10, 10, &golden_frame_);
  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
  VerifyFramesAreEqual(dst_frame_, golden_frame_);

  // Mono -> stereo.
  SetMonoFrame(0, &dst_frame_);
  SetStereoFrame(10, 20, &src_frame_);
  SetMonoFrame(15, &golden_frame_);
  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
  VerifyFramesAreEqual(golden_frame_, dst_frame_);
}

TEST_F(UtilityTest, RemixAndResampleSucceeds) {
  const int kSampleRates[] = {8000,  11025, 16000, 22050,
                              32000, 44100, 48000, 96000};
  const int kSrcChannels[] = {1, 2, 4};
  const int kDstChannels[] = {1, 2};

  for (int src_rate : kSampleRates) {
    for (int dst_rate : kSampleRates) {
      for (size_t src_channels : kSrcChannels) {
        for (size_t dst_channels : kDstChannels) {
          RunResampleTest(src_channels, src_rate, dst_channels, dst_rate);
        }
      }
    }
  }
}

}  // namespace
}  // namespace voe
}  // namespace webrtc