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
This commit is contained in:
andrew@webrtc.org
parent
5d608955cf
commit
2c29c2eae2
@ -8,39 +8,179 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/base/checks.h"
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#include "webrtc/common_audio/audio_converter.h"
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#include <cstring>
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#include "webrtc/base/checks.h"
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#include "webrtc/base/safe_conversions.h"
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#include "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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#include "webrtc/system_wrappers/interface/scoped_vector.h"
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using rtc::checked_cast;
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namespace webrtc {
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namespace {
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void DownmixToMono(const float* const* src,
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int src_channels,
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int frames,
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float* dst) {
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DCHECK_GT(src_channels, 0);
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for (int i = 0; i < frames; ++i) {
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float sum = 0;
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for (int j = 0; j < src_channels; ++j)
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sum += src[j][i];
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dst[i] = sum / src_channels;
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class CopyConverter : public AudioConverter {
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public:
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CopyConverter(int src_channels, int src_frames, int dst_channels,
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int dst_frames)
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: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
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~CopyConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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CheckSizes(src_size, dst_capacity);
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if (src != dst) {
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for (int i = 0; i < src_channels(); ++i)
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std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
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}
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}
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};
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class UpmixConverter : public AudioConverter {
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public:
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UpmixConverter(int src_channels, int src_frames, int dst_channels,
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int dst_frames)
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: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
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~UpmixConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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CheckSizes(src_size, dst_capacity);
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for (int i = 0; i < dst_frames(); ++i) {
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const float value = src[0][i];
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for (int j = 0; j < dst_channels(); ++j)
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dst[j][i] = value;
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}
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}
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};
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class DownmixConverter : public AudioConverter {
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public:
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DownmixConverter(int src_channels, int src_frames, int dst_channels,
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int dst_frames)
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: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
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}
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~DownmixConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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CheckSizes(src_size, dst_capacity);
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float* dst_mono = dst[0];
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for (int i = 0; i < src_frames(); ++i) {
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float sum = 0;
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for (int j = 0; j < src_channels(); ++j)
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sum += src[j][i];
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dst_mono[i] = sum / src_channels();
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}
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}
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};
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class ResampleConverter : public AudioConverter {
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public:
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ResampleConverter(int src_channels, int src_frames, int dst_channels,
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int dst_frames)
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: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
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resamplers_.reserve(src_channels);
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for (int i = 0; i < src_channels; ++i)
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resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
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}
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~ResampleConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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CheckSizes(src_size, dst_capacity);
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for (size_t i = 0; i < resamplers_.size(); ++i)
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resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames());
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}
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private:
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ScopedVector<PushSincResampler> resamplers_;
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};
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// Apply a vector of converters in serial, in the order given. At least two
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// converters must be provided.
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class CompositionConverter : public AudioConverter {
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public:
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CompositionConverter(ScopedVector<AudioConverter> converters)
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: converters_(converters.Pass()) {
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CHECK_GE(converters_.size(), 2u);
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// We need an intermediate buffer after every converter.
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for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
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buffers_.push_back(new ChannelBuffer<float>((*it)->dst_frames(),
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(*it)->dst_channels()));
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}
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~CompositionConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
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buffers_.front()->size());
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for (size_t i = 2; i < converters_.size(); ++i) {
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auto src_buffer = buffers_[i - 2];
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auto dst_buffer = buffers_[i - 1];
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converters_[i]->Convert(src_buffer->channels(),
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src_buffer->size(),
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dst_buffer->channels(),
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dst_buffer->size());
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}
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converters_.back()->Convert(buffers_.back()->channels(),
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buffers_.back()->size(), dst, dst_capacity);
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}
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private:
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ScopedVector<AudioConverter> converters_;
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ScopedVector<ChannelBuffer<float>> buffers_;
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};
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scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
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int src_frames,
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int dst_channels,
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int dst_frames) {
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scoped_ptr<AudioConverter> sp;
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if (src_channels > dst_channels) {
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if (src_frames != dst_frames) {
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ScopedVector<AudioConverter> converters;
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converters.push_back(new DownmixConverter(src_channels, src_frames,
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dst_channels, src_frames));
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converters.push_back(new ResampleConverter(dst_channels, src_frames,
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dst_channels, dst_frames));
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sp.reset(new CompositionConverter(converters.Pass()));
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} else {
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sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
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dst_frames));
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}
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} else if (src_channels < dst_channels) {
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if (src_frames != dst_frames) {
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ScopedVector<AudioConverter> converters;
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converters.push_back(new ResampleConverter(src_channels, src_frames,
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src_channels, dst_frames));
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converters.push_back(new UpmixConverter(src_channels, dst_frames,
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dst_channels, dst_frames));
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sp.reset(new CompositionConverter(converters.Pass()));
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} else {
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sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
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dst_frames));
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}
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} else if (src_frames != dst_frames) {
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sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
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dst_frames));
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} else {
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sp.reset(new CopyConverter(src_channels, src_frames, dst_channels,
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dst_frames));
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}
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return sp.Pass();
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}
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void UpmixFromMono(const float* src,
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int dst_channels,
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int frames,
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float* const* dst) {
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DCHECK_GT(dst_channels, 0);
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for (int i = 0; i < frames; ++i) {
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float value = src[i];
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for (int j = 0; j < dst_channels; ++j)
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dst[j][i] = value;
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}
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}
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} // namespace
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// For CompositionConverter.
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AudioConverter::AudioConverter()
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: src_channels_(0),
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src_frames_(0),
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dst_channels_(0),
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dst_frames_(0) {}
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AudioConverter::AudioConverter(int src_channels, int src_frames,
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int dst_channels, int dst_frames)
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@ -49,62 +189,11 @@ AudioConverter::AudioConverter(int src_channels, int src_frames,
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dst_channels_(dst_channels),
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dst_frames_(dst_frames) {
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CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
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const int resample_channels = std::min(src_channels, dst_channels);
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// Prepare buffers as needed for intermediate stages.
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if (dst_channels < src_channels)
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downmix_buffer_.reset(new ChannelBuffer<float>(src_frames,
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resample_channels));
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if (src_frames != dst_frames) {
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resamplers_.reserve(resample_channels);
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for (int i = 0; i < resample_channels; ++i)
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resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
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}
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}
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void AudioConverter::Convert(const float* const* src,
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int src_channels,
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int src_frames,
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int dst_channels,
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int dst_frames,
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float* const* dst) {
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DCHECK_EQ(src_channels_, src_channels);
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DCHECK_EQ(src_frames_, src_frames);
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DCHECK_EQ(dst_channels_, dst_channels);
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DCHECK_EQ(dst_frames_, dst_frames);;
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if (src_channels == dst_channels && src_frames == dst_frames) {
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// Shortcut copy.
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if (src != dst) {
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for (int i = 0; i < src_channels; ++i)
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memcpy(dst[i], src[i], dst_frames * sizeof(*dst[i]));
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}
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return;
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}
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const float* const* src_ptr = src;
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if (dst_channels < src_channels) {
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float* const* dst_ptr = dst;
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if (src_frames != dst_frames) {
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// Downmix to a buffer for subsequent resampling.
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DCHECK_EQ(downmix_buffer_->num_channels(), dst_channels);
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DCHECK_EQ(downmix_buffer_->num_frames(), src_frames);
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dst_ptr = downmix_buffer_->channels();
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}
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DownmixToMono(src, src_channels, src_frames, dst_ptr[0]);
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src_ptr = dst_ptr;
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}
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if (src_frames != dst_frames) {
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for (size_t i = 0; i < resamplers_.size(); ++i)
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resamplers_[i]->Resample(src_ptr[i], src_frames, dst[i], dst_frames);
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src_ptr = dst;
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}
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if (dst_channels > src_channels)
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UpmixFromMono(src_ptr[0], dst_channels, dst_frames, dst);
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void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
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CHECK_EQ(src_size, checked_cast<size_t>(src_channels() * src_frames()));
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CHECK_GE(dst_capacity, checked_cast<size_t>(dst_channels() * dst_frames()));
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}
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} // namespace webrtc
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@ -11,16 +11,11 @@
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#ifndef WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
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#define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
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// TODO(ajm): Move channel buffer to common_audio.
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/scoped_vector.h"
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namespace webrtc {
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class PushSincResampler;
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// Format conversion (remixing and resampling) for audio. Only simple remixing
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// conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
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// upmix from mono (i.e. |src_channels == 1|).
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@ -29,23 +24,37 @@ class PushSincResampler;
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// the number of frames is equivalent to specifying the sample rates.
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class AudioConverter {
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public:
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AudioConverter(int src_channels, int src_frames,
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int dst_channels, int dst_frames);
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// Returns a new AudioConverter, which will use the supplied format for its
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// lifetime. Caller is responsible for the memory.
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static scoped_ptr<AudioConverter> Create(int src_channels, int src_frames,
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int dst_channels, int dst_frames);
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virtual ~AudioConverter() {};
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void Convert(const float* const* src,
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int src_channels,
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int src_frames,
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int dst_channels,
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int dst_frames,
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float* const* dest);
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// Convert |src|, containing |src_size| samples, to |dst|, having a sample
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// capacity of |dst_capacity|. Both point to a series of buffers containing
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// the samples for each channel. The sizes must correspond to the format
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// passed to Create().
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virtual void Convert(const float* const* src, size_t src_size,
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float* const* dst, size_t dst_capacity) = 0;
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int src_channels() const { return src_channels_; }
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int src_frames() const { return src_frames_; }
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int dst_channels() const { return dst_channels_; }
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int dst_frames() const { return dst_frames_; }
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protected:
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AudioConverter();
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AudioConverter(int src_channels, int src_frames, int dst_channels,
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int dst_frames);
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// Helper to CHECK that inputs are correctly sized.
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void CheckSizes(size_t src_size, size_t dst_capacity) const;
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private:
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const int src_channels_;
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const int src_frames_;
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const int dst_channels_;
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const int dst_frames_;
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scoped_ptr<ChannelBuffer<float>> downmix_buffer_;
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ScopedVector<PushSincResampler> resamplers_;
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DISALLOW_COPY_AND_ASSIGN(AudioConverter);
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};
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@ -8,14 +8,14 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <math.h>
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#include <cmath>
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#include <algorithm>
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#include <vector>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_audio/audio_converter.h"
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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#include "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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@ -63,6 +63,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
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mean += ref.channels()[i][j];
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}
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}
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const int length = ref.num_channels() * (ref.num_frames() - delay);
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mse /= length;
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variance /= length;
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@ -70,7 +71,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
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variance -= mean * mean;
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float snr = 100; // We assign 100 dB to the zero-error case.
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if (mse > 0)
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snr = 10 * log10(variance / mse);
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snr = 10 * std::log10(variance / mse);
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if (snr > best_snr) {
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best_snr = snr;
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best_delay = delay;
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@ -127,9 +128,11 @@ void RunAudioConverterTest(int src_channels,
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printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later.
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src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
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AudioConverter converter(src_channels, src_frames, dst_channels, dst_frames);
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converter.Convert(src_buffer->channels(), src_channels, src_frames,
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dst_channels, dst_frames, dst_buffer->channels());
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scoped_ptr<AudioConverter> converter =
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AudioConverter::Create(src_channels, src_frames, dst_channels,
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dst_frames);
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converter->Convert(src_buffer->channels(), src_buffer->size(),
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dst_buffer->channels(), dst_buffer->size());
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EXPECT_LT(43.f,
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ComputeSNR(*ref_buffer.get(), *dst_buffer.get(), delay_frames));
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@ -8,21 +8,21 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/common_audio/include/audio_util.h"
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#include <assert.h>
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#include <string.h>
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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#include <cstring>
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#include "webrtc/base/checks.h"
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#include "webrtc/common_audio/include/audio_util.h"
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namespace webrtc {
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PushSincResampler::PushSincResampler(int source_frames, int destination_frames)
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: resampler_(new SincResampler(source_frames * 1.0 / destination_frames,
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source_frames,
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this)),
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source_ptr_(NULL),
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source_ptr_int_(NULL),
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source_ptr_(nullptr),
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source_ptr_int_(nullptr),
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destination_frames_(destination_frames),
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first_pass_(true),
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source_available_(0) {}
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@ -38,10 +38,10 @@ int PushSincResampler::Resample(const int16_t* source,
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float_buffer_.reset(new float[destination_frames_]);
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source_ptr_int_ = source;
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// Pass NULL as the float source to have Run() read from the int16 source.
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Resample(NULL, source_length, float_buffer_.get(), destination_frames_);
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// Pass nullptr as the float source to have Run() read from the int16 source.
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Resample(nullptr, source_length, float_buffer_.get(), destination_frames_);
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FloatS16ToS16(float_buffer_.get(), destination_frames_, destination);
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source_ptr_int_ = NULL;
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source_ptr_int_ = nullptr;
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return destination_frames_;
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}
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@ -49,8 +49,8 @@ int PushSincResampler::Resample(const float* source,
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int source_length,
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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 @@ 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]);
|
||||
|
||||
@ -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.
|
||||
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_;
|
||||
|
||||
@ -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 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),
|
||||
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 @@ INSTANTIATE_TEST_CASE_P(
|
||||
// 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
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user