1 files changed, 276 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/remix_resample_unittest.cc b/third_party/libwebrtc/audio/remix_resample_unittest.cc
new file mode 100644
index 0000000000..31dcfac1fe
--- /dev/null
+++ b/third_party/libwebrtc/audio/remix_resample_unittest.cc
@@ -0,0 +1,276 @@
+/*
+ *  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