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
diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
index 2549393..05262f8 100644
--- a/webrtc/common_audio/audio_converter.cc
+++ b/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,
- int src_channels,
- int frames,
- float* dst) {
- DCHECK_GT(src_channels, 0);
- for (int i = 0; i < frames; ++i) {
- float sum = 0;
- for (int j = 0; j < src_channels; ++j)
- sum += src[j][i];
- dst[i] = sum / src_channels;
+class CopyConverter : public AudioConverter {
+ public:
+ CopyConverter(int src_channels, int src_frames, int dst_channels,
+ int dst_frames)
+ : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
+ ~CopyConverter() override {};
+
+ void Convert(const float* const* src, size_t src_size, float* const* dst,
+ size_t dst_capacity) override {
+ 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]));
+ }
}
+};
+
+class UpmixConverter : public AudioConverter {
+ public:
+ UpmixConverter(int src_channels, int src_frames, int dst_channels,
+ int dst_frames)
+ : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
+ ~UpmixConverter() 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 (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();
}
-void UpmixFromMono(const float* src,
- int dst_channels,
- int frames,
- float* const* dst) {
- DCHECK_GT(dst_channels, 0);
- for (int i = 0; i < frames; ++i) {
- float value = src[i];
- for (int j = 0; j < dst_channels; ++j)
- dst[j][i] = value;
- }
-}
-
-} // 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 @@
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,
- int src_channels,
- int src_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);
+void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
+ CHECK_EQ(src_size, checked_cast<size_t>(src_channels() * src_frames()));
+ CHECK_GE(dst_capacity, checked_cast<size_t>(dst_channels() * dst_frames()));
}
} // namespace webrtc
diff --git a/webrtc/common_audio/audio_converter.h b/webrtc/common_audio/audio_converter.h
index 564eda1..3873827 100644
--- a/webrtc/common_audio/audio_converter.h
+++ b/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 @@
// the number of frames is equivalent to specifying the sample rates.
class AudioConverter {
public:
- AudioConverter(int src_channels, int src_frames,
- int dst_channels, int dst_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,
- int src_channels,
- int src_frames,
- int dst_channels,
- int dst_frames,
- float* const* dest);
+ // Convert |src|, containing |src_size| samples, to |dst|, having a sample
+ // capacity of |dst_capacity|. Both point to a series of buffers containing
+ // the samples for each channel. The sizes must correspond to the format
+ // passed to Create().
+ virtual void Convert(const float* const* src, size_t src_size,
+ 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);
};
diff --git a/webrtc/common_audio/audio_converter_unittest.cc b/webrtc/common_audio/audio_converter_unittest.cc
index 205bd98..03c0d47 100644
--- a/webrtc/common_audio/audio_converter_unittest.cc
+++ b/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 @@
mean += ref.channels()[i][j];
}
}
+
const int length = ref.num_channels() * (ref.num_frames() - delay);
mse /= length;
variance /= length;
@@ -70,7 +71,7 @@
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 @@
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);
- converter.Convert(src_buffer->channels(), src_channels, src_frames,
- dst_channels, dst_frames, dst_buffer->channels());
+ scoped_ptr<AudioConverter> converter =
+ AudioConverter::Create(src_channels, src_frames, dst_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));
diff --git a/webrtc/common_audio/resampler/push_sinc_resampler.cc b/webrtc/common_audio/resampler/push_sinc_resampler.cc
index 49e2e12..7d37202 100644
--- a/webrtc/common_audio/resampler/push_sinc_resampler.cc
+++ b/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_int_(NULL),
+ source_ptr_(nullptr),
+ source_ptr_int_(nullptr),
destination_frames_(destination_frames),
first_pass_(true),
source_available_(0) {}
@@ -38,10 +38,10 @@
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.
- Resample(NULL, source_length, float_buffer_.get(), destination_frames_);
+ // Pass nullptr as the float source to have Run() read from the int16 source.
+ 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 source_length,
float* destination,
int destination_capacity) {
- assert(source_length == resampler_->request_frames());
- assert(destination_capacity >= destination_frames_);
+ CHECK_EQ(source_length, resampler_->request_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 @@
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]);
diff --git a/webrtc/common_audio/resampler/push_sinc_resampler.h b/webrtc/common_audio/resampler/push_sinc_resampler.h
index df724e2..e68a2fb 100644
--- a/webrtc/common_audio/resampler/push_sinc_resampler.h
+++ b/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 @@
float* destination,
int destination_capacity);
- // Implements SincResamplerCallback.
- virtual void Run(int frames, float* destination) OVERRIDE;
-
- SincResampler* get_resampler_for_testing() { return resampler_.get(); }
+ // Delay due to the filter kernel. Essentially, the time after which an input
+ // sample will appear in the resampled output.
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_;
diff --git a/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc b/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
index 90ac0cf..c39a7a4 100644
--- a/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
+++ b/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;
-class PushSincResamplerTest
- : public testing::TestWithParam<PushSincResamplerTestData> {
+// Almost all conversions have an RMS error of around -14 dbFS.
+const double kResamplingRMSError = -14.42;
+
+// 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())),
- output_rate_(std::tr1::get<1>(GetParam())),
- rms_error_(std::tr1::get<2>(GetParam())),
- low_freq_error_(std::tr1::get<3>(GetParam())) {
+ : input_rate_(::testing::get<0>(GetParam())),
+ output_rate_(::testing::get<1>(GetParam())),
+ rms_error_(::testing::get<2>(GetParam())),
+ low_freq_error_(::testing::get<3>(GetParam())) {
}
- virtual ~PushSincResamplerTest() {}
+ ~PushSincResamplerTest() override {}
protected:
void ResampleBenchmarkTest(bool int_format);
@@ -47,7 +59,7 @@
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 @@
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, 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 @@
// these rates in any case (for the same reason).
// To 44.1kHz
- std::tr1::make_tuple(8000, 44100, kResamplingRMSError, -62.73),
- std::tr1::make_tuple(16000, 44100, kResamplingRMSError, -62.54),
- std::tr1::make_tuple(32000, 44100, kResamplingRMSError, -63.32),
- std::tr1::make_tuple(44100, 44100, kResamplingRMSError, -73.53),
- std::tr1::make_tuple(48000, 44100, -15.01, -64.04),
- std::tr1::make_tuple(96000, 44100, -18.49, -25.51),
- std::tr1::make_tuple(192000, 44100, -20.50, -13.31),
+ ::testing::make_tuple(8000, 44100, kResamplingRMSError, -62.73),
+ ::testing::make_tuple(16000, 44100, kResamplingRMSError, -62.54),
+ ::testing::make_tuple(32000, 44100, kResamplingRMSError, -63.32),
+ ::testing::make_tuple(44100, 44100, kResamplingRMSError, -73.53),
+ ::testing::make_tuple(48000, 44100, -15.01, -64.04),
+ ::testing::make_tuple(96000, 44100, -18.49, -25.51),
+ ::testing::make_tuple(192000, 44100, -20.50, -13.31),
// To 48kHz
- std::tr1::make_tuple(8000, 48000, kResamplingRMSError, -63.43),
- std::tr1::make_tuple(16000, 48000, kResamplingRMSError, -63.96),
- std::tr1::make_tuple(32000, 48000, kResamplingRMSError, -64.04),
- std::tr1::make_tuple(44100, 48000, kResamplingRMSError, -62.63),
- std::tr1::make_tuple(48000, 48000, kResamplingRMSError, -73.52),
- std::tr1::make_tuple(96000, 48000, -18.40, -28.44),
- std::tr1::make_tuple(192000, 48000, -20.43, -14.11),
+ ::testing::make_tuple(8000, 48000, kResamplingRMSError, -63.43),
+ ::testing::make_tuple(16000, 48000, kResamplingRMSError, -63.96),
+ ::testing::make_tuple(32000, 48000, kResamplingRMSError, -64.04),
+ ::testing::make_tuple(44100, 48000, kResamplingRMSError, -62.63),
+ ::testing::make_tuple(48000, 48000, kResamplingRMSError, -73.52),
+ ::testing::make_tuple(96000, 48000, -18.40, -28.44),
+ ::testing::make_tuple(192000, 48000, -20.43, -14.11),
// To 96kHz
- std::tr1::make_tuple(8000, 96000, kResamplingRMSError, -63.19),
- std::tr1::make_tuple(16000, 96000, kResamplingRMSError, -63.39),
- std::tr1::make_tuple(32000, 96000, kResamplingRMSError, -63.95),
- std::tr1::make_tuple(44100, 96000, kResamplingRMSError, -62.63),
- std::tr1::make_tuple(48000, 96000, kResamplingRMSError, -73.52),
- std::tr1::make_tuple(96000, 96000, kResamplingRMSError, -73.52),
- std::tr1::make_tuple(192000, 96000, kResamplingRMSError, -28.41),
+ ::testing::make_tuple(8000, 96000, kResamplingRMSError, -63.19),
+ ::testing::make_tuple(16000, 96000, kResamplingRMSError, -63.39),
+ ::testing::make_tuple(32000, 96000, kResamplingRMSError, -63.95),
+ ::testing::make_tuple(44100, 96000, kResamplingRMSError, -62.63),
+ ::testing::make_tuple(48000, 96000, kResamplingRMSError, -73.52),
+ ::testing::make_tuple(96000, 96000, kResamplingRMSError, -73.52),
+ ::testing::make_tuple(192000, 96000, kResamplingRMSError, -28.41),
// To 192kHz
- std::tr1::make_tuple(8000, 192000, kResamplingRMSError, -63.10),
- std::tr1::make_tuple(16000, 192000, kResamplingRMSError, -63.14),
- std::tr1::make_tuple(32000, 192000, kResamplingRMSError, -63.38),
- std::tr1::make_tuple(44100, 192000, kResamplingRMSError, -62.63),
- std::tr1::make_tuple(48000, 192000, kResamplingRMSError, -73.44),
- std::tr1::make_tuple(96000, 192000, kResamplingRMSError, -73.52),
- std::tr1::make_tuple(192000, 192000, kResamplingRMSError, -73.52),
+ ::testing::make_tuple(8000, 192000, kResamplingRMSError, -63.10),
+ ::testing::make_tuple(16000, 192000, kResamplingRMSError, -63.14),
+ ::testing::make_tuple(32000, 192000, kResamplingRMSError, -63.38),
+ ::testing::make_tuple(44100, 192000, kResamplingRMSError, -62.63),
+ ::testing::make_tuple(48000, 192000, kResamplingRMSError, -73.44),
+ ::testing::make_tuple(96000, 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 @@
// practice anyway.
// To 8 kHz
- std::tr1::make_tuple(8000, 8000, kResamplingRMSError, -75.50),
- std::tr1::make_tuple(16000, 8000, -18.56, -28.79),
- std::tr1::make_tuple(32000, 8000, -20.36, -14.13),
- std::tr1::make_tuple(44100, 8000, -21.00, -11.39),
- std::tr1::make_tuple(48000, 8000, -20.96, -11.04),
+ ::testing::make_tuple(8000, 8000, kResamplingRMSError, -75.50),
+ ::testing::make_tuple(16000, 8000, -18.56, -28.79),
+ ::testing::make_tuple(32000, 8000, -20.36, -14.13),
+ ::testing::make_tuple(44100, 8000, -21.00, -11.39),
+ ::testing::make_tuple(48000, 8000, -20.96, -11.04),
// To 16 kHz
- std::tr1::make_tuple(8000, 16000, kResamplingRMSError, -70.30),
- std::tr1::make_tuple(16000, 16000, kResamplingRMSError, -75.51),
- std::tr1::make_tuple(32000, 16000, -18.48, -28.59),
- std::tr1::make_tuple(44100, 16000, -19.30, -19.67),
- std::tr1::make_tuple(48000, 16000, -19.81, -18.11),
- std::tr1::make_tuple(96000, 16000, -20.95, -10.96),
+ ::testing::make_tuple(8000, 16000, kResamplingRMSError, -70.30),
+ ::testing::make_tuple(16000, 16000, kResamplingRMSError, -75.51),
+ ::testing::make_tuple(32000, 16000, -18.48, -28.59),
+ ::testing::make_tuple(44100, 16000, -19.30, -19.67),
+ ::testing::make_tuple(48000, 16000, -19.81, -18.11),
+ ::testing::make_tuple(96000, 16000, -20.95, -10.96),
// To 32 kHz
- std::tr1::make_tuple(8000, 32000, kResamplingRMSError, -70.30),
- std::tr1::make_tuple(16000, 32000, kResamplingRMSError, -75.51),
- std::tr1::make_tuple(32000, 32000, kResamplingRMSError, -75.51),
- std::tr1::make_tuple(44100, 32000, -16.44, -51.10),
- std::tr1::make_tuple(48000, 32000, -16.90, -44.03),
- std::tr1::make_tuple(96000, 32000, -19.61, -18.04),
- std::tr1::make_tuple(192000, 32000, -21.02, -10.94)));
+ ::testing::make_tuple(8000, 32000, kResamplingRMSError, -70.30),
+ ::testing::make_tuple(16000, 32000, kResamplingRMSError, -75.51),
+ ::testing::make_tuple(32000, 32000, kResamplingRMSError, -75.51),
+ ::testing::make_tuple(44100, 32000, -16.44, -51.10),
+ ::testing::make_tuple(48000, 32000, -16.90, -44.03),
+ ::testing::make_tuple(96000, 32000, -19.61, -18.04),
+ ::testing::make_tuple(192000, 32000, -21.02, -10.94)));
} // namespace webrtc
