Bug 901633 - Part 14 - Add testing for our audio processing functions. r=jesup

dom/media/compiledtest/TestAudioPacketizer.cpp

@@ -5,6 +5,8 @@
 
 #include <stdint.h>
 #include <assert.h>
+#define _USE_MATH_DEFINES
+#include <math.h>
 #include "../AudioPacketizer.h"
 
 using namespace mozilla;
@@ -13,7 +15,7 @@ template<typename T>
 class AutoBuffer
 {
 public:
-  AutoBuffer(size_t aLength)
+  explicit AutoBuffer(size_t aLength)
   {
     mStorage = new T[aLength];
   }

dom/media/compiledtest/TestAudioSegment.cpp

@@ -0,0 +1,268 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioSegment.h"
+#include <assert.h>
+#include <iostream>
+
+using namespace mozilla;
+
+namespace mozilla {
+uint32_t
+GetAudioChannelsSuperset(uint32_t aChannels1, uint32_t aChannels2)
+{
+  return std::max(aChannels1, aChannels2);
+}
+}
+
+/* Helper function to give us the maximum and minimum value that don't clip,
+ * for a given sample format (integer or floating-point). */
+template<typename T>
+T GetLowValue();
+
+template<typename T>
+T GetHighValue();
+
+template<typename T>
+T GetSilentValue();
+
+template<>
+float GetLowValue<float>() {
+  return -1.0;
+}
+
+template<>
+int16_t GetLowValue<short>() {
+  return -INT16_MAX;
+}
+
+template<>
+float GetHighValue<float>() {
+  return 1.0;
+}
+
+template<>
+int16_t GetHighValue<short>() {
+  return INT16_MAX;
+}
+
+template<>
+float GetSilentValue() {
+  return 0.0;
+}
+
+template<>
+int16_t GetSilentValue() {
+  return 0;
+}
+
+
+// Get an array of planar audio buffers that has the inverse of the index of the
+// channel (1-indexed) as samples.
+template<typename T>
+const T* const* GetPlanarChannelArray(size_t aChannels, size_t aSize)
+{
+  T** channels = new T*[aChannels];
+  for (size_t c = 0; c < aChannels; c++) {
+    channels[c] = new T[aSize];
+    for (size_t i = 0; i < aSize; i++) {
+      channels[c][i] = FloatToAudioSample<T>(1. / (c + 1));
+    }
+  }
+  return channels;
+}
+
+template<typename T>
+void DeletePlanarChannelsArray(const T* const* aArrays, size_t aChannels)
+{
+  for (size_t channel = 0; channel < aChannels; channel++) {
+    delete [] aArrays[channel];
+  }
+  delete [] aArrays;
+}
+
+template<typename T>
+T** GetPlanarArray(size_t aChannels, size_t aSize)
+{
+  T** channels = new T*[aChannels];
+  for (size_t c = 0; c < aChannels; c++) {
+    channels[c] = new T[aSize];
+    for (size_t i = 0; i < aSize; i++) {
+      channels[c][i] = 0.0f;
+    }
+  }
+  return channels;
+}
+
+template<typename T>
+void DeletePlanarArray(T** aArrays, size_t aChannels)
+{
+  for (size_t channel = 0; channel < aChannels; channel++) {
+    delete [] aArrays[channel];
+  }
+  delete [] aArrays;
+}
+
+// Get an array of audio samples that have the inverse of the index of the
+// channel (1-indexed) as samples.
+template<typename T>
+const T* GetInterleavedChannelArray(size_t aChannels, size_t aSize)
+{
+  size_t sampleCount = aChannels * aSize;
+  T* samples = new T[sampleCount];
+  for (size_t i = 0; i < sampleCount; i++) {
+    uint32_t channel = (i % aChannels) + 1;
+    samples[i] = FloatToAudioSample<T>(1. / channel);
+  }
+  return samples;
+}
+
+template<typename T>
+void DeleteInterleavedChannelArray(const T* aArray)
+{
+  delete [] aArray;
+}
+
+bool FuzzyEqual(float aLhs, float aRhs) {
+  return std::abs(aLhs - aRhs) < 0.01;
+}
+
+template<typename SrcT, typename DstT>
+void TestInterleaveAndConvert()
+{
+  size_t arraySize = 1024;
+  size_t maxChannels = 8; // 7.1
+  for (uint32_t channels = 1; channels < maxChannels; channels++) {
+    const SrcT* const* src = GetPlanarChannelArray<SrcT>(channels, arraySize);
+    DstT* dst = new DstT[channels * arraySize];
+
+    InterleaveAndConvertBuffer(src, arraySize, 1.0, channels, dst);
+
+    uint32_t channelIndex = 0;
+    for (size_t i = 0; i < arraySize * channels; i++) {
+      assert(FuzzyEqual(dst[i],
+                        FloatToAudioSample<DstT>(1. / (channelIndex + 1))));
+      channelIndex++;
+      channelIndex %= channels;
+    }
+
+    DeletePlanarChannelsArray(src, channels);
+    delete [] dst;
+  }
+}
+
+template<typename SrcT, typename DstT>
+void TestDeinterleaveAndConvert()
+{
+  size_t arraySize = 1024;
+  size_t maxChannels = 8; // 7.1
+  for (uint32_t channels = 1; channels < maxChannels; channels++) {
+    const SrcT* src = GetInterleavedChannelArray<SrcT>(channels, arraySize);
+    DstT** dst = GetPlanarArray<DstT>(channels, arraySize);
+
+    DeinterleaveAndConvertBuffer(src, arraySize, channels, dst);
+
+    for (size_t channel = 0; channel < channels; channel++) {
+      for (size_t i = 0; i < arraySize; i++) {
+        assert(FuzzyEqual(dst[channel][i],
+                          FloatToAudioSample<DstT>(1. / (channel + 1))));
+      }
+    }
+
+    DeleteInterleavedChannelArray(src);
+    DeletePlanarArray(dst, channels);
+  }
+}
+
+uint8_t gSilence[4096] = {0};
+
+template<typename T>
+T* SilentChannel()
+{
+  return reinterpret_cast<T*>(gSilence);
+}
+
+template<typename T>
+void TestUpmixStereo()
+{
+  size_t arraySize = 1024;
+  nsTArray<T*> channels;
+  nsTArray<const T*> channelsptr;
+
+  channels.SetLength(1);
+  channelsptr.SetLength(1);
+
+  channels[0] = new T[arraySize];
+
+  for (size_t i = 0; i < arraySize; i++) {
+    channels[0][i] = GetHighValue<T>();
+  }
+  channelsptr[0] = channels[0];
+
+  AudioChannelsUpMix(&channelsptr, 2, ::SilentChannel<T>());
+
+  for (size_t channel = 0; channel < 2; channel++) {
+    for (size_t i = 0; i < arraySize; i++) {
+      if (channelsptr[channel][i] != GetHighValue<T>()) {
+        assert(false);
+      }
+    }
+  }
+  assert(true);
+  delete channels[0];
+}
+
+template<typename T>
+void TestDownmixStereo()
+{
+  const size_t arraySize = 1024;
+  nsTArray<const T*> inputptr;
+  nsTArray<T*> input;
+  T** output;
+
+  output = new T*[1];
+  output[0] = new T[arraySize];
+
+  input.SetLength(2);
+  inputptr.SetLength(2);
+
+  for (size_t channel = 0; channel < input.Length(); channel++) {
+    input[channel] = new T[arraySize];
+    for (size_t i = 0; i < arraySize; i++) {
+      input[channel][i] = channel == 0 ? GetLowValue<T>() : GetHighValue<T>();
+    }
+    inputptr[channel] = input[channel];
+  }
+
+  AudioChannelsDownMix(inputptr, output, 1, arraySize);
+
+  for (size_t i = 0; i < arraySize; i++) {
+    if (output[0][i] != GetSilentValue<T>()) {
+      assert(false);
+    }
+  }
+  assert(true);
+
+  delete output[0];
+  delete output;
+}
+
+int main(int argc, char* argv[]) {
+  TestInterleaveAndConvert<float, float>();
+  TestInterleaveAndConvert<float, int16_t>();
+  TestInterleaveAndConvert<int16_t, float>();
+  TestInterleaveAndConvert<int16_t, int16_t>();
+  TestDeinterleaveAndConvert<float, float>();
+  TestDeinterleaveAndConvert<float, int16_t>();
+  TestDeinterleaveAndConvert<int16_t, float>();
+  TestDeinterleaveAndConvert<int16_t, int16_t>();
+  TestUpmixStereo<float>();
+  TestUpmixStereo<int16_t>();
+  TestDownmixStereo<float>();
+  TestDownmixStereo<int16_t>();
+
+  return 0;
+}
+

dom/media/compiledtest/moz.build

@@ -7,7 +7,8 @@
 GeckoCppUnitTests([
     'TestAudioBuffers',
     'TestAudioMixer',
-    'TestAudioPacketizer'
+    'TestAudioPacketizer',
+    'TestAudioSegment'
 ])
 
 LOCAL_INCLUDES += [