C++ readability review for ajm.

As part of the review, refactored AudioConverter into internal derived
classes, each focused on one type of conversion. A factory method
returns the correct converter (or chain of converters, via
CompositionConverter).

BUG=b/18938079
R=rojer@google.com

Review URL: https://webrtc-codereview.appspot.com/35699004

Cr-Commit-Position: refs/heads/master@{#8322}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8322 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/common_audio/audio_converter.cc

@@ -8,39 +8,179 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/audio_converter.h"
+
+#include <cstring>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
+#include "webrtc/system_wrappers/interface/scoped_vector.h"
+
+using rtc::checked_cast;
 
 namespace webrtc {
-namespace {
 
-void DownmixToMono(const float* const* src,
+class CopyConverter : public AudioConverter {
-                   int src_channels,
+ public:
-                   int frames,
+  CopyConverter(int src_channels, int src_frames, int dst_channels,
-                   float* dst) {
+                int dst_frames)
-  DCHECK_GT(src_channels, 0);
+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
-  for (int i = 0; i < frames; ++i) {
+  ~CopyConverter() override {};
-    float sum = 0;
+
-    for (int j = 0; j < src_channels; ++j)
+  void Convert(const float* const* src, size_t src_size, float* const* dst,
-      sum += src[j][i];
+               size_t dst_capacity) override {
-    dst[i] = sum / src_channels;
+    CheckSizes(src_size, dst_capacity);
+    if (src != dst) {
+      for (int i = 0; i < src_channels(); ++i)
+        std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
     }
-}
+  }
+};
 
-void UpmixFromMono(const float* src,
+class UpmixConverter : public AudioConverter {
-                   int dst_channels,
+ public:
-                   int frames,
+  UpmixConverter(int src_channels, int src_frames, int dst_channels,
-                   float* const* dst) {
+                 int dst_frames)
-  DCHECK_GT(dst_channels, 0);
+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
-  for (int i = 0; i < frames; ++i) {
+  ~UpmixConverter() override {};
-    float value = src[i];
+
-    for (int j = 0; j < dst_channels; ++j)
+  void Convert(const float* const* src, size_t src_size, float* const* dst,
+               size_t dst_capacity) override {
+    CheckSizes(src_size, dst_capacity);
+    for (int i = 0; i < dst_frames(); ++i) {
+      const float value = src[0][i];
+      for (int j = 0; j < dst_channels(); ++j)
         dst[j][i] = value;
     }
+  }
+};
+
+class DownmixConverter : public AudioConverter {
+ public:
+  DownmixConverter(int src_channels, int src_frames, int dst_channels,
+                   int dst_frames)
+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
+  }
+  ~DownmixConverter() override {};
+
+  void Convert(const float* const* src, size_t src_size, float* const* dst,
+               size_t dst_capacity) override {
+    CheckSizes(src_size, dst_capacity);
+    float* dst_mono = dst[0];
+    for (int i = 0; i < src_frames(); ++i) {
+      float sum = 0;
+      for (int j = 0; j < src_channels(); ++j)
+        sum += src[j][i];
+      dst_mono[i] = sum / src_channels();
+    }
+  }
+};
+
+class ResampleConverter : public AudioConverter {
+ public:
+  ResampleConverter(int src_channels, int src_frames, int dst_channels,
+                    int dst_frames)
+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
+    resamplers_.reserve(src_channels);
+    for (int i = 0; i < src_channels; ++i)
+      resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
+  }
+  ~ResampleConverter() override {};
+
+  void Convert(const float* const* src, size_t src_size, float* const* dst,
+               size_t dst_capacity) override {
+    CheckSizes(src_size, dst_capacity);
+    for (size_t i = 0; i < resamplers_.size(); ++i)
+      resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames());
+  }
+
+ private:
+  ScopedVector<PushSincResampler> resamplers_;
+};
+
+// Apply a vector of converters in serial, in the order given. At least two
+// converters must be provided.
+class CompositionConverter : public AudioConverter {
+ public:
+  CompositionConverter(ScopedVector<AudioConverter> converters)
+      : converters_(converters.Pass()) {
+    CHECK_GE(converters_.size(), 2u);
+    // We need an intermediate buffer after every converter.
+    for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
+      buffers_.push_back(new ChannelBuffer<float>((*it)->dst_frames(),
+                                                  (*it)->dst_channels()));
+  }
+  ~CompositionConverter() override {};
+
+  void Convert(const float* const* src, size_t src_size, float* const* dst,
+               size_t dst_capacity) override {
+    converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
+                                 buffers_.front()->size());
+    for (size_t i = 2; i < converters_.size(); ++i) {
+      auto src_buffer = buffers_[i - 2];
+      auto dst_buffer = buffers_[i - 1];
+      converters_[i]->Convert(src_buffer->channels(),
+                              src_buffer->size(),
+                              dst_buffer->channels(),
+                              dst_buffer->size());
+    }
+    converters_.back()->Convert(buffers_.back()->channels(),
+                                buffers_.back()->size(), dst, dst_capacity);
+  }
+
+ private:
+  ScopedVector<AudioConverter> converters_;
+  ScopedVector<ChannelBuffer<float>> buffers_;
+};
+
+scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
+                                                  int src_frames,
+                                                  int dst_channels,
+                                                  int dst_frames) {
+  scoped_ptr<AudioConverter> sp;
+  if (src_channels > dst_channels) {
+    if (src_frames != dst_frames) {
+      ScopedVector<AudioConverter> converters;
+      converters.push_back(new DownmixConverter(src_channels, src_frames,
+                                                dst_channels, src_frames));
+      converters.push_back(new ResampleConverter(dst_channels, src_frames,
+                                                 dst_channels, dst_frames));
+      sp.reset(new CompositionConverter(converters.Pass()));
+    } else {
+      sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
+                                    dst_frames));
+    }
+  } else if (src_channels < dst_channels) {
+    if (src_frames != dst_frames) {
+      ScopedVector<AudioConverter> converters;
+      converters.push_back(new ResampleConverter(src_channels, src_frames,
+                                                 src_channels, dst_frames));
+      converters.push_back(new UpmixConverter(src_channels, dst_frames,
+                                              dst_channels, dst_frames));
+      sp.reset(new CompositionConverter(converters.Pass()));
+    } else {
+      sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
+                                  dst_frames));
+    }
+  } else if (src_frames != dst_frames) {
+    sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
+                                   dst_frames));
+  } else {
+    sp.reset(new CopyConverter(src_channels, src_frames, dst_channels,
+                               dst_frames));
+  }
+
+  return sp.Pass();
 }
 
-}  // namespace
+// For CompositionConverter.
+AudioConverter::AudioConverter()
+    : src_channels_(0),
+      src_frames_(0),
+      dst_channels_(0),
+      dst_frames_(0) {}
 
 AudioConverter::AudioConverter(int src_channels, int src_frames,
                                int dst_channels, int dst_frames)
@@ -49,62 +189,11 @@ AudioConverter::AudioConverter(int src_channels, int src_frames,
       dst_channels_(dst_channels),
       dst_frames_(dst_frames) {
   CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
-  const int resample_channels = std::min(src_channels, dst_channels);
-
-  // Prepare buffers as needed for intermediate stages.
-  if (dst_channels < src_channels)
-    downmix_buffer_.reset(new ChannelBuffer<float>(src_frames,
-                                                   resample_channels));
-
-  if (src_frames != dst_frames) {
-    resamplers_.reserve(resample_channels);
-    for (int i = 0; i < resample_channels; ++i)
-      resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
-  }
 }
 
-void AudioConverter::Convert(const float* const* src,
+void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
-                             int src_channels,
+  CHECK_EQ(src_size, checked_cast<size_t>(src_channels() * src_frames()));
-                             int src_frames,
+  CHECK_GE(dst_capacity, checked_cast<size_t>(dst_channels() * dst_frames()));
-                             int dst_channels,
-                             int dst_frames,
-                             float* const* dst) {
-  DCHECK_EQ(src_channels_, src_channels);
-  DCHECK_EQ(src_frames_, src_frames);
-  DCHECK_EQ(dst_channels_, dst_channels);
-  DCHECK_EQ(dst_frames_, dst_frames);;
-
-  if (src_channels == dst_channels && src_frames == dst_frames) {
-    // Shortcut copy.
-    if (src != dst) {
-      for (int i = 0; i < src_channels; ++i)
-        memcpy(dst[i], src[i], dst_frames * sizeof(*dst[i]));
-    }
-    return;
-  }
-
-  const float* const* src_ptr = src;
-  if (dst_channels < src_channels) {
-    float* const* dst_ptr = dst;
-    if (src_frames != dst_frames) {
-      // Downmix to a buffer for subsequent resampling.
-      DCHECK_EQ(downmix_buffer_->num_channels(), dst_channels);
-      DCHECK_EQ(downmix_buffer_->num_frames(), src_frames);
-      dst_ptr = downmix_buffer_->channels();
-    }
-
-    DownmixToMono(src, src_channels, src_frames, dst_ptr[0]);
-    src_ptr = dst_ptr;
-  }
-
-  if (src_frames != dst_frames) {
-    for (size_t i = 0; i < resamplers_.size(); ++i)
-      resamplers_[i]->Resample(src_ptr[i], src_frames, dst[i], dst_frames);
-    src_ptr = dst;
-  }
-
-  if (dst_channels > src_channels)
-    UpmixFromMono(src_ptr[0], dst_channels, dst_frames, dst);
 }
 
 }  // namespace webrtc

webrtc/common_audio/audio_converter.h

@@ -11,16 +11,11 @@
 #ifndef WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
 #define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
 
-// TODO(ajm): Move channel buffer to common_audio.
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
 
 namespace webrtc {
 
-class PushSincResampler;
-
 // Format conversion (remixing and resampling) for audio. Only simple remixing
 // conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
 // upmix from mono (i.e. |src_channels == 1|).
@@ -29,23 +24,37 @@ class PushSincResampler;
 // the number of frames is equivalent to specifying the sample rates.
 class AudioConverter {
  public:
-  AudioConverter(int src_channels, int src_frames,
+  // Returns a new AudioConverter, which will use the supplied format for its
+  // lifetime. Caller is responsible for the memory.
+  static scoped_ptr<AudioConverter> Create(int src_channels, int src_frames,
                                            int dst_channels, int dst_frames);
+  virtual ~AudioConverter() {};
 
-  void Convert(const float* const* src,
+  // Convert |src|, containing |src_size| samples, to |dst|, having a sample
-               int src_channels,
+  // capacity of |dst_capacity|. Both point to a series of buffers containing
-               int src_frames,
+  // the samples for each channel. The sizes must correspond to the format
-               int dst_channels,
+  // passed to Create().
-               int dst_frames,
+  virtual void Convert(const float* const* src, size_t src_size,
-               float* const* dest);
+                       float* const* dst, size_t dst_capacity) = 0;
+
+  int src_channels() const { return src_channels_; }
+  int src_frames() const { return src_frames_; }
+  int dst_channels() const { return dst_channels_; }
+  int dst_frames() const { return dst_frames_; }
+
+ protected:
+  AudioConverter();
+  AudioConverter(int src_channels, int src_frames, int dst_channels,
+                 int dst_frames);
+
+  // Helper to CHECK that inputs are correctly sized.
+  void CheckSizes(size_t src_size, size_t dst_capacity) const;
 
  private:
   const int src_channels_;
   const int src_frames_;
   const int dst_channels_;
   const int dst_frames_;
-  scoped_ptr<ChannelBuffer<float>> downmix_buffer_;
-  ScopedVector<PushSincResampler> resamplers_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioConverter);
 };

webrtc/common_audio/audio_converter_unittest.cc

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <math.h>
+#include <cmath>
 #include <algorithm>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_audio/audio_converter.h"
-#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
@@ -63,6 +63,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
         mean += ref.channels()[i][j];
       }
     }
+
     const int length = ref.num_channels() * (ref.num_frames() - delay);
     mse /= length;
     variance /= length;
@@ -70,7 +71,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
     variance -= mean * mean;
     float snr = 100;  // We assign 100 dB to the zero-error case.
     if (mse > 0)
-      snr = 10 * log10(variance / mse);
+      snr = 10 * std::log10(variance / mse);
     if (snr > best_snr) {
       best_snr = snr;
       best_delay = delay;
@@ -127,9 +128,11 @@ void RunAudioConverterTest(int src_channels,
   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);
 
-  AudioConverter converter(src_channels, src_frames, dst_channels, dst_frames);
+  scoped_ptr<AudioConverter> converter =
-  converter.Convert(src_buffer->channels(), src_channels, src_frames,
+      AudioConverter::Create(src_channels, src_frames, dst_channels,
-                    dst_channels, dst_frames, dst_buffer->channels());
+                             dst_frames);
+  converter->Convert(src_buffer->channels(), src_buffer->size(),
+                     dst_buffer->channels(), dst_buffer->size());
 
   EXPECT_LT(43.f,
             ComputeSNR(*ref_buffer.get(), *dst_buffer.get(), delay_frames));

webrtc/common_audio/resampler/push_sinc_resampler.cc

@@ -8,21 +8,21 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/common_audio/include/audio_util.h"
-
-#include <assert.h>
-#include <string.h>
-
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 
+#include <cstring>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/include/audio_util.h"
+
 namespace webrtc {
 
 PushSincResampler::PushSincResampler(int source_frames, int destination_frames)
     : resampler_(new SincResampler(source_frames * 1.0 / destination_frames,
                                    source_frames,
                                    this)),
-      source_ptr_(NULL),
+      source_ptr_(nullptr),
-      source_ptr_int_(NULL),
+      source_ptr_int_(nullptr),
       destination_frames_(destination_frames),
       first_pass_(true),
       source_available_(0) {}
@@ -38,10 +38,10 @@ int PushSincResampler::Resample(const int16_t* source,
     float_buffer_.reset(new float[destination_frames_]);
 
   source_ptr_int_ = source;
-  // Pass NULL as the float source to have Run() read from the int16 source.
+  // Pass nullptr as the float source to have Run() read from the int16 source.
-  Resample(NULL, source_length, float_buffer_.get(), destination_frames_);
+  Resample(nullptr, source_length, float_buffer_.get(), destination_frames_);
   FloatS16ToS16(float_buffer_.get(), destination_frames_, destination);
-  source_ptr_int_ = NULL;
+  source_ptr_int_ = nullptr;
   return destination_frames_;
 }
 
@@ -49,8 +49,8 @@ int PushSincResampler::Resample(const float* source,
                                 int source_length,
                                 float* destination,
                                 int destination_capacity) {
-  assert(source_length == resampler_->request_frames());
+  CHECK_EQ(source_length, resampler_->request_frames());
-  assert(destination_capacity >= destination_frames_);
+  CHECK_GE(destination_capacity, destination_frames_);
   // Cache the source pointer. Calling Resample() will immediately trigger
   // the Run() callback whereupon we provide the cached value.
   source_ptr_ = source;
@@ -73,25 +73,25 @@ int PushSincResampler::Resample(const float* source,
     resampler_->Resample(resampler_->ChunkSize(), destination);
 
   resampler_->Resample(destination_frames_, destination);
-  source_ptr_ = NULL;
+  source_ptr_ = nullptr;
   return destination_frames_;
 }
 
 void PushSincResampler::Run(int frames, float* destination) {
   // Ensure we are only asked for the available samples. This would fail if
   // Run() was triggered more than once per Resample() call.
-  assert(source_available_ == frames);
+  CHECK_EQ(source_available_, frames);
 
   if (first_pass_) {
     // Provide dummy input on the first pass, the output of which will be
     // discarded, as described in Resample().
-    memset(destination, 0, frames * sizeof(float));
+    std::memset(destination, 0, frames * sizeof(*destination));
     first_pass_ = false;
     return;
   }
 
   if (source_ptr_) {
-    memcpy(destination, source_ptr_, frames * sizeof(float));
+    std::memcpy(destination, source_ptr_, frames * sizeof(*destination));
   } else {
     for (int i = 0; i < frames; ++i)
       destination[i] = static_cast<float>(source_ptr_int_[i]);

webrtc/common_audio/resampler/push_sinc_resampler.h

@@ -19,14 +19,16 @@
 namespace webrtc {
 
 // A thin wrapper over SincResampler to provide a push-based interface as
-// required by WebRTC.
+// required by WebRTC. SincResampler uses a pull-based interface, and will
+// use SincResamplerCallback::Run() to request data upon a call to Resample().
+// These Run() calls will happen on the same thread Resample() is called on.
 class PushSincResampler : public SincResamplerCallback {
  public:
   // Provide the size of the source and destination blocks in samples. These
   // must correspond to the same time duration (typically 10 ms) as the sample
   // ratio is inferred from them.
   PushSincResampler(int source_frames, int destination_frames);
-  virtual ~PushSincResampler();
+  ~PushSincResampler() override;
 
   // Perform the resampling. |source_frames| must always equal the
   // |source_frames| provided at construction. |destination_capacity| must be
@@ -40,15 +42,20 @@ class PushSincResampler : public SincResamplerCallback {
                float* destination,
                int destination_capacity);
 
-  // Implements SincResamplerCallback.
+  // Delay due to the filter kernel. Essentially, the time after which an input
-  virtual void Run(int frames, float* destination) OVERRIDE;
+  // sample will appear in the resampled output.
-
-  SincResampler* get_resampler_for_testing() { return resampler_.get(); }
   static float AlgorithmicDelaySeconds(int source_rate_hz) {
     return 1.f / source_rate_hz * SincResampler::kKernelSize / 2;
   }
 
+ protected:
+  // Implements SincResamplerCallback.
+  void Run(int frames, float* destination) override;
+
  private:
+  friend class PushSincResamplerTest;
+  SincResampler* get_resampler_for_testing() { return resampler_.get(); }
+
   scoped_ptr<SincResampler> resampler_;
   scoped_ptr<float[]> float_buffer_;
   const float* source_ptr_;

webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc

@@ -8,7 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <math.h>
+#include <cmath>
+#include <cstring>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ -20,19 +21,30 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+namespace {
 
-typedef std::tr1::tuple<int, int, double, double> PushSincResamplerTestData;
+// Almost all conversions have an RMS error of around -14 dbFS.
-class PushSincResamplerTest
+const double kResamplingRMSError = -14.42;
-    : public testing::TestWithParam<PushSincResamplerTestData> {
+
+// Used to convert errors to dbFS.
+template <typename T>
+T DBFS(T x) {
+  return 20 * std::log10(x);
+}
+
+}  // namespace
+
+class PushSincResamplerTest : public ::testing::TestWithParam<
+    ::testing::tuple<int, int, double, double>> {
  public:
   PushSincResamplerTest()
-      : input_rate_(std::tr1::get<0>(GetParam())),
+      : input_rate_(::testing::get<0>(GetParam())),
-        output_rate_(std::tr1::get<1>(GetParam())),
+        output_rate_(::testing::get<1>(GetParam())),
-        rms_error_(std::tr1::get<2>(GetParam())),
+        rms_error_(::testing::get<2>(GetParam())),
-        low_freq_error_(std::tr1::get<3>(GetParam())) {
+        low_freq_error_(::testing::get<3>(GetParam())) {
   }
 
-  virtual ~PushSincResamplerTest() {}
+  ~PushSincResamplerTest() override {}
 
  protected:
   void ResampleBenchmarkTest(bool int_format);
@@ -47,7 +59,7 @@ class PushSincResamplerTest
 class ZeroSource : public SincResamplerCallback {
  public:
   void Run(int frames, float* destination) {
-    memset(destination, 0, sizeof(float) * frames);
+    std::memset(destination, 0, sizeof(float) * frames);
   }
 };
 
@@ -216,8 +228,6 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
 
   double rms_error = sqrt(sum_of_squares / output_samples);
 
-  // Convert each error to dbFS.
-  #define DBFS(x) 20 * log10(x)
   rms_error = DBFS(rms_error);
   // In order to keep the thresholds in this test identical to SincResamplerTest
   // we must account for the quantization error introduced by truncating from
@@ -241,15 +251,12 @@ TEST_P(PushSincResamplerTest, ResampleInt) { ResampleTest(true); }
 
 TEST_P(PushSincResamplerTest, ResampleFloat) { ResampleTest(false); }
 
-// Almost all conversions have an RMS error of around -14 dbFS.
-static const double kResamplingRMSError = -14.42;
-
 // Thresholds chosen arbitrarily based on what each resampling reported during
 // testing.  All thresholds are in dbFS, http://en.wikipedia.org/wiki/DBFS.
 INSTANTIATE_TEST_CASE_P(
     PushSincResamplerTest,
     PushSincResamplerTest,
-    testing::Values(
+    ::testing::Values(
         // First run through the rates tested in SincResamplerTest. The
         // thresholds are identical.
         //
@@ -258,40 +265,40 @@ INSTANTIATE_TEST_CASE_P(
         // these rates in any case (for the same reason).
 
         // To 44.1kHz
-        std::tr1::make_tuple(8000, 44100, kResamplingRMSError, -62.73),
+        ::testing::make_tuple(8000, 44100, kResamplingRMSError, -62.73),
-        std::tr1::make_tuple(16000, 44100, kResamplingRMSError, -62.54),
+        ::testing::make_tuple(16000, 44100, kResamplingRMSError, -62.54),
-        std::tr1::make_tuple(32000, 44100, kResamplingRMSError, -63.32),
+        ::testing::make_tuple(32000, 44100, kResamplingRMSError, -63.32),
-        std::tr1::make_tuple(44100, 44100, kResamplingRMSError, -73.53),
+        ::testing::make_tuple(44100, 44100, kResamplingRMSError, -73.53),
-        std::tr1::make_tuple(48000, 44100, -15.01, -64.04),
+        ::testing::make_tuple(48000, 44100, -15.01, -64.04),
-        std::tr1::make_tuple(96000, 44100, -18.49, -25.51),
+        ::testing::make_tuple(96000, 44100, -18.49, -25.51),
-        std::tr1::make_tuple(192000, 44100, -20.50, -13.31),
+        ::testing::make_tuple(192000, 44100, -20.50, -13.31),
 
         // To 48kHz
-        std::tr1::make_tuple(8000, 48000, kResamplingRMSError, -63.43),
+        ::testing::make_tuple(8000, 48000, kResamplingRMSError, -63.43),
-        std::tr1::make_tuple(16000, 48000, kResamplingRMSError, -63.96),
+        ::testing::make_tuple(16000, 48000, kResamplingRMSError, -63.96),
-        std::tr1::make_tuple(32000, 48000, kResamplingRMSError, -64.04),
+        ::testing::make_tuple(32000, 48000, kResamplingRMSError, -64.04),
-        std::tr1::make_tuple(44100, 48000, kResamplingRMSError, -62.63),
+        ::testing::make_tuple(44100, 48000, kResamplingRMSError, -62.63),
-        std::tr1::make_tuple(48000, 48000, kResamplingRMSError, -73.52),
+        ::testing::make_tuple(48000, 48000, kResamplingRMSError, -73.52),
-        std::tr1::make_tuple(96000, 48000, -18.40, -28.44),
+        ::testing::make_tuple(96000, 48000, -18.40, -28.44),
-        std::tr1::make_tuple(192000, 48000, -20.43, -14.11),
+        ::testing::make_tuple(192000, 48000, -20.43, -14.11),
 
         // To 96kHz
-        std::tr1::make_tuple(8000, 96000, kResamplingRMSError, -63.19),
+        ::testing::make_tuple(8000, 96000, kResamplingRMSError, -63.19),
-        std::tr1::make_tuple(16000, 96000, kResamplingRMSError, -63.39),
+        ::testing::make_tuple(16000, 96000, kResamplingRMSError, -63.39),
-        std::tr1::make_tuple(32000, 96000, kResamplingRMSError, -63.95),
+        ::testing::make_tuple(32000, 96000, kResamplingRMSError, -63.95),
-        std::tr1::make_tuple(44100, 96000, kResamplingRMSError, -62.63),
+        ::testing::make_tuple(44100, 96000, kResamplingRMSError, -62.63),
-        std::tr1::make_tuple(48000, 96000, kResamplingRMSError, -73.52),
+        ::testing::make_tuple(48000, 96000, kResamplingRMSError, -73.52),
-        std::tr1::make_tuple(96000, 96000, kResamplingRMSError, -73.52),
+        ::testing::make_tuple(96000, 96000, kResamplingRMSError, -73.52),
-        std::tr1::make_tuple(192000, 96000, kResamplingRMSError, -28.41),
+        ::testing::make_tuple(192000, 96000, kResamplingRMSError, -28.41),
 
         // To 192kHz
-        std::tr1::make_tuple(8000, 192000, kResamplingRMSError, -63.10),
+        ::testing::make_tuple(8000, 192000, kResamplingRMSError, -63.10),
-        std::tr1::make_tuple(16000, 192000, kResamplingRMSError, -63.14),
+        ::testing::make_tuple(16000, 192000, kResamplingRMSError, -63.14),
-        std::tr1::make_tuple(32000, 192000, kResamplingRMSError, -63.38),
+        ::testing::make_tuple(32000, 192000, kResamplingRMSError, -63.38),
-        std::tr1::make_tuple(44100, 192000, kResamplingRMSError, -62.63),
+        ::testing::make_tuple(44100, 192000, kResamplingRMSError, -62.63),
-        std::tr1::make_tuple(48000, 192000, kResamplingRMSError, -73.44),
+        ::testing::make_tuple(48000, 192000, kResamplingRMSError, -73.44),
-        std::tr1::make_tuple(96000, 192000, kResamplingRMSError, -73.52),
+        ::testing::make_tuple(96000, 192000, kResamplingRMSError, -73.52),
-        std::tr1::make_tuple(192000, 192000, kResamplingRMSError, -73.52),
+        ::testing::make_tuple(192000, 192000, kResamplingRMSError, -73.52),
 
         // Next run through some additional cases interesting for WebRTC.
         // We skip some extreme downsampled cases (192 -> {8, 16}, 96 -> 8)
@@ -300,27 +307,27 @@ INSTANTIATE_TEST_CASE_P(
         // practice anyway.
 
         // To 8 kHz
-        std::tr1::make_tuple(8000, 8000, kResamplingRMSError, -75.50),
+        ::testing::make_tuple(8000, 8000, kResamplingRMSError, -75.50),
-        std::tr1::make_tuple(16000, 8000, -18.56, -28.79),
+        ::testing::make_tuple(16000, 8000, -18.56, -28.79),
-        std::tr1::make_tuple(32000, 8000, -20.36, -14.13),
+        ::testing::make_tuple(32000, 8000, -20.36, -14.13),
-        std::tr1::make_tuple(44100, 8000, -21.00, -11.39),
+        ::testing::make_tuple(44100, 8000, -21.00, -11.39),
-        std::tr1::make_tuple(48000, 8000, -20.96, -11.04),
+        ::testing::make_tuple(48000, 8000, -20.96, -11.04),
 
         // To 16 kHz
-        std::tr1::make_tuple(8000, 16000, kResamplingRMSError, -70.30),
+        ::testing::make_tuple(8000, 16000, kResamplingRMSError, -70.30),
-        std::tr1::make_tuple(16000, 16000, kResamplingRMSError, -75.51),
+        ::testing::make_tuple(16000, 16000, kResamplingRMSError, -75.51),
-        std::tr1::make_tuple(32000, 16000, -18.48, -28.59),
+        ::testing::make_tuple(32000, 16000, -18.48, -28.59),
-        std::tr1::make_tuple(44100, 16000, -19.30, -19.67),
+        ::testing::make_tuple(44100, 16000, -19.30, -19.67),
-        std::tr1::make_tuple(48000, 16000, -19.81, -18.11),
+        ::testing::make_tuple(48000, 16000, -19.81, -18.11),
-        std::tr1::make_tuple(96000, 16000, -20.95, -10.96),
+        ::testing::make_tuple(96000, 16000, -20.95, -10.96),
 
         // To 32 kHz
-        std::tr1::make_tuple(8000, 32000, kResamplingRMSError, -70.30),
+        ::testing::make_tuple(8000, 32000, kResamplingRMSError, -70.30),
-        std::tr1::make_tuple(16000, 32000, kResamplingRMSError, -75.51),
+        ::testing::make_tuple(16000, 32000, kResamplingRMSError, -75.51),
-        std::tr1::make_tuple(32000, 32000, kResamplingRMSError, -75.51),
+        ::testing::make_tuple(32000, 32000, kResamplingRMSError, -75.51),
-        std::tr1::make_tuple(44100, 32000, -16.44, -51.10),
+        ::testing::make_tuple(44100, 32000, -16.44, -51.10),
-        std::tr1::make_tuple(48000, 32000, -16.90, -44.03),
+        ::testing::make_tuple(48000, 32000, -16.90, -44.03),
-        std::tr1::make_tuple(96000, 32000, -19.61, -18.04),
+        ::testing::make_tuple(96000, 32000, -19.61, -18.04),
-        std::tr1::make_tuple(192000, 32000, -21.02, -10.94)));
+        ::testing::make_tuple(192000, 32000, -21.02, -10.94)));
 
 }  // namespace webrtc