| /* |
| * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ |
| #define MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ |
| |
| #include <math.h> |
| |
| #include <iterator> |
| #include <limits> |
| #include <memory> |
| #include <sstream> // no-presubmit-check TODO(webrtc:8982) |
| #include <string> |
| #include <vector> |
| |
| #include "common_audio/channel_buffer.h" |
| #include "common_audio/wav_file.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "rtc_base/constructor_magic.h" |
| |
| namespace webrtc { |
| |
| static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError; |
| #define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr)) |
| |
| class RawFile final { |
| public: |
| explicit RawFile(const std::string& filename); |
| ~RawFile(); |
| |
| void WriteSamples(const int16_t* samples, size_t num_samples); |
| void WriteSamples(const float* samples, size_t num_samples); |
| |
| private: |
| FILE* file_handle_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(RawFile); |
| }; |
| |
| // Encapsulates samples and metadata for an integer frame. |
| struct Int16FrameData { |
| // Max data size that matches the data size of the AudioFrame class, providing |
| // storage for 8 channels of 96 kHz data. |
| static const int kMaxDataSizeSamples = 7680; |
| |
| Int16FrameData() { |
| sample_rate_hz = 0; |
| num_channels = 0; |
| samples_per_channel = 0; |
| data.fill(0); |
| } |
| |
| void CopyFrom(const Int16FrameData& src) { |
| samples_per_channel = src.samples_per_channel; |
| sample_rate_hz = src.sample_rate_hz; |
| num_channels = src.num_channels; |
| |
| const size_t length = samples_per_channel * num_channels; |
| RTC_CHECK_LE(length, kMaxDataSizeSamples); |
| memcpy(data.data(), src.data.data(), sizeof(int16_t) * length); |
| } |
| std::array<int16_t, kMaxDataSizeSamples> data; |
| int32_t sample_rate_hz; |
| size_t num_channels; |
| size_t samples_per_channel; |
| }; |
| |
| // Reads ChannelBuffers from a provided WavReader. |
| class ChannelBufferWavReader final { |
| public: |
| explicit ChannelBufferWavReader(std::unique_ptr<WavReader> file); |
| ~ChannelBufferWavReader(); |
| |
| // Reads data from the file according to the `buffer` format. Returns false if |
| // a full buffer can't be read from the file. |
| bool Read(ChannelBuffer<float>* buffer); |
| |
| private: |
| std::unique_ptr<WavReader> file_; |
| std::vector<float> interleaved_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader); |
| }; |
| |
| // Writes ChannelBuffers to a provided WavWriter. |
| class ChannelBufferWavWriter final { |
| public: |
| explicit ChannelBufferWavWriter(std::unique_ptr<WavWriter> file); |
| ~ChannelBufferWavWriter(); |
| |
| void Write(const ChannelBuffer<float>& buffer); |
| |
| private: |
| std::unique_ptr<WavWriter> file_; |
| std::vector<float> interleaved_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter); |
| }; |
| |
| // Takes a pointer to a vector. Allows appending the samples of channel buffers |
| // to the given vector, by interleaving the samples and converting them to float |
| // S16. |
| class ChannelBufferVectorWriter final { |
| public: |
| explicit ChannelBufferVectorWriter(std::vector<float>* output); |
| ChannelBufferVectorWriter(const ChannelBufferVectorWriter&) = delete; |
| ChannelBufferVectorWriter& operator=(const ChannelBufferVectorWriter&) = |
| delete; |
| ~ChannelBufferVectorWriter(); |
| |
| // Creates an interleaved copy of `buffer`, converts the samples to float S16 |
| // and appends the result to output_. |
| void Write(const ChannelBuffer<float>& buffer); |
| |
| private: |
| std::vector<float> interleaved_buffer_; |
| std::vector<float>* output_; |
| }; |
| |
| void WriteIntData(const int16_t* data, |
| size_t length, |
| WavWriter* wav_file, |
| RawFile* raw_file); |
| |
| void WriteFloatData(const float* const* data, |
| size_t samples_per_channel, |
| size_t num_channels, |
| WavWriter* wav_file, |
| RawFile* raw_file); |
| |
| // Exits on failure; do not use in unit tests. |
| FILE* OpenFile(const std::string& filename, const char* mode); |
| |
| size_t SamplesFromRate(int rate); |
| |
| void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz); |
| |
| template <typename T> |
| void SetContainerFormat(int sample_rate_hz, |
| size_t num_channels, |
| Int16FrameData* frame, |
| std::unique_ptr<ChannelBuffer<T> >* cb) { |
| SetFrameSampleRate(frame, sample_rate_hz); |
| frame->num_channels = num_channels; |
| cb->reset(new ChannelBuffer<T>(frame->samples_per_channel, num_channels)); |
| } |
| |
| AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels); |
| |
| template <typename T> |
| float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) { |
| float mse = 0; |
| float mean = 0; |
| *variance = 0; |
| for (size_t i = 0; i < length; ++i) { |
| T error = ref[i] - test[i]; |
| mse += error * error; |
| *variance += ref[i] * ref[i]; |
| mean += ref[i]; |
| } |
| mse /= length; |
| *variance /= length; |
| mean /= length; |
| *variance -= mean * mean; |
| |
| float snr = 100; // We assign 100 dB to the zero-error case. |
| if (mse > 0) |
| snr = 10 * log10(*variance / mse); |
| return snr; |
| } |
| |
| // Returns a vector<T> parsed from whitespace delimited values in to_parse, |
| // or an empty vector if the string could not be parsed. |
| template <typename T> |
| std::vector<T> ParseList(const std::string& to_parse) { |
| std::vector<T> values; |
| |
| std::istringstream str(to_parse); |
| std::copy( |
| std::istream_iterator<T>(str), // no-presubmit-check TODO(webrtc:8982) |
| std::istream_iterator<T>(), // no-presubmit-check TODO(webrtc:8982) |
| std::back_inserter(values)); |
| |
| return values; |
| } |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ |