1 files changed, 263 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/voice_engine/utility_unittest.cc b/chromium/third_party/webrtc/voice_engine/utility_unittest.cc
new file mode 100644
index 00000000000..8f7efa87f65
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+++ b/chromium/third_party/webrtc/voice_engine/utility_unittest.cc
@@ -0,0 +1,263 @@
+/*
+ *  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 <math.h>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_audio/resampler/include/push_resampler.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/voice_engine/utility.h"
+#include "webrtc/voice_engine/voice_engine_defines.h"
+
+namespace webrtc {
+namespace voe {
+namespace {
+
+enum FunctionToTest {
+  TestRemixAndResample,
+  TestDownConvertToCodecFormat
+};
+
+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,
+                       FunctionToTest function);
+
+  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(AudioFrame* frame, float data, int sample_rate_hz) {
+  memset(frame->data_, 0, sizeof(frame->data_));
+  frame->num_channels_ = 1;
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = sample_rate_hz / 100;
+  for (int i = 0; i < frame->samples_per_channel_; i++) {
+    frame->data_[i] = data * i;
+  }
+}
+
+// Keep the existing sample rate.
+void SetMonoFrame(AudioFrame* frame, float data) {
+  SetMonoFrame(frame, data, frame->sample_rate_hz_);
+}
+
+// Sets the signal value to increase by |left| and |right| with every sample in
+// each channel respectively.
+void SetStereoFrame(AudioFrame* frame, float left, float right,
+                    int sample_rate_hz) {
+  memset(frame->data_, 0, sizeof(frame->data_));
+  frame->num_channels_ = 2;
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = sample_rate_hz / 100;
+  for (int i = 0; i < frame->samples_per_channel_; i++) {
+    frame->data_[i * 2] = left * i;
+    frame->data_[i * 2 + 1] = right * i;
+  }
+}
+
+// Keep the existing sample rate.
+void SetStereoFrame(AudioFrame* frame, float left, float right) {
+  SetStereoFrame(frame, left, right, frame->sample_rate_hz_);
+}
+
+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,
+                 int max_delay) {
+  VerifyParams(ref_frame, test_frame);
+  float best_snr = 0;
+  int best_delay = 0;
+  for (int delay = 0; delay <= max_delay; delay++) {
+    float mse = 0;
+    float variance = 0;
+    for (int 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 * log10(variance / mse);
+    if (snr > best_snr) {
+      best_snr = snr;
+      best_delay = delay;
+    }
+  }
+  printf("SNR=%.1f dB at delay=%d\n", best_snr, best_delay);
+  return best_snr;
+}
+
+void VerifyFramesAreEqual(const AudioFrame& ref_frame,
+                          const AudioFrame& test_frame) {
+  VerifyParams(ref_frame, test_frame);
+  for (int 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,
+                                  FunctionToTest function) {
+  PushResampler<int16_t> resampler;  // Create a new one with every test.
+  const int16_t kSrcLeft = 30;  // Shouldn't overflow for any used sample rate.
+  const int16_t kSrcRight = 15;
+  const float resampling_factor = (1.0 * src_sample_rate_hz) /
+      dst_sample_rate_hz;
+  const float dst_left = resampling_factor * kSrcLeft;
+  const float dst_right = resampling_factor * kSrcRight;
+  const float dst_mono = (dst_left + dst_right) / 2;
+  if (src_channels == 1)
+    SetMonoFrame(&src_frame_, kSrcLeft, src_sample_rate_hz);
+  else
+    SetStereoFrame(&src_frame_, kSrcLeft, kSrcRight, src_sample_rate_hz);
+
+  if (dst_channels == 1) {
+    SetMonoFrame(&dst_frame_, 0, dst_sample_rate_hz);
+    if (src_channels == 1)
+      SetMonoFrame(&golden_frame_, dst_left, dst_sample_rate_hz);
+    else
+      SetMonoFrame(&golden_frame_, dst_mono, dst_sample_rate_hz);
+  } else {
+    SetStereoFrame(&dst_frame_, 0, 0, dst_sample_rate_hz);
+    if (src_channels == 1)
+      SetStereoFrame(&golden_frame_, dst_left, dst_left, dst_sample_rate_hz);
+    else
+      SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
+  }
+
+  // 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 int kInputKernelDelaySamples = 16;
+  const int max_delay = 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);
+  if (function == TestRemixAndResample) {
+    RemixAndResample(src_frame_, &resampler, &dst_frame_);
+  } else {
+    int16_t mono_buffer[kMaxMonoDataSizeSamples];
+    DownConvertToCodecFormat(src_frame_.data_,
+                             src_frame_.samples_per_channel_,
+                             src_frame_.num_channels_,
+                             src_frame_.sample_rate_hz_,
+                             dst_frame_.num_channels_,
+                             dst_frame_.sample_rate_hz_,
+                             mono_buffer,
+                             &resampler,
+                             &dst_frame_);
+  }
+
+  if (src_sample_rate_hz == 96000 && dst_sample_rate_hz == 8000) {
+    // 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(&src_frame_, 10, 10);
+  SetStereoFrame(&dst_frame_, 0, 0);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(src_frame_, dst_frame_);
+
+  // Mono -> mono.
+  SetMonoFrame(&src_frame_, 20);
+  SetMonoFrame(&dst_frame_, 0);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(src_frame_, dst_frame_);
+}
+
+TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) {
+  // Stereo -> mono.
+  SetStereoFrame(&dst_frame_, 0, 0);
+  SetMonoFrame(&src_frame_, 10);
+  SetStereoFrame(&golden_frame_, 10, 10);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(dst_frame_, golden_frame_);
+
+  // Mono -> stereo.
+  SetMonoFrame(&dst_frame_, 0);
+  SetStereoFrame(&src_frame_, 10, 20);
+  SetMonoFrame(&golden_frame_, 15);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(golden_frame_, dst_frame_);
+}
+
+TEST_F(UtilityTest, RemixAndResampleSucceeds) {
+  const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
+  const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
+  const int kChannels[] = {1, 2};
+  const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
+  for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
+    for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
+      for (int src_channel = 0; src_channel < kChannelsSize; src_channel++) {
+        for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
+          RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
+                          kChannels[dst_channel], kSampleRates[dst_rate],
+                          TestRemixAndResample);
+        }
+      }
+    }
+  }
+}
+
+TEST_F(UtilityTest, ConvertToCodecFormatSucceeds) {
+  const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
+  const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
+  const int kChannels[] = {1, 2};
+  const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
+  for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
+    for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
+      for (int src_channel = 0; src_channel < kChannelsSize; src_channel++) {
+        for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
+          if (dst_rate <= src_rate && dst_channel <= src_channel) {
+            RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
+                            kChannels[src_channel], kSampleRates[dst_rate],
+                            TestDownConvertToCodecFormat);
+          }
+        }
+      }
+    }
+  }
+}
+
+}  // namespace
+}  // namespace voe
+}  // namespace webrtc