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| 1 /* | 1 /* |
| 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ |
| 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ | 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ |
| 13 | 13 |
| 14 #include <math.h> | 14 #include <math.h> |
| 15 #include <iterator> | 15 #include <iterator> |
| 16 #include <limits> | 16 #include <limits> |
| 17 #include <memory> |
| 17 #include <string> | 18 #include <string> |
| 18 #include <vector> | 19 #include <vector> |
| 19 | 20 |
| 20 #include "webrtc/base/constructormagic.h" | 21 #include "webrtc/base/constructormagic.h" |
| 21 #include "webrtc/base/scoped_ptr.h" | |
| 22 #include "webrtc/common_audio/channel_buffer.h" | 22 #include "webrtc/common_audio/channel_buffer.h" |
| 23 #include "webrtc/common_audio/wav_file.h" | 23 #include "webrtc/common_audio/wav_file.h" |
| 24 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 24 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| 25 #include "webrtc/modules/include/module_common_types.h" | 25 #include "webrtc/modules/include/module_common_types.h" |
| 26 | 26 |
| 27 namespace webrtc { | 27 namespace webrtc { |
| 28 | 28 |
| 29 static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError; | 29 static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError; |
| 30 #define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr)) | 30 #define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr)) |
| 31 | 31 |
| 32 class RawFile final { | 32 class RawFile final { |
| 33 public: | 33 public: |
| 34 explicit RawFile(const std::string& filename); | 34 explicit RawFile(const std::string& filename); |
| 35 ~RawFile(); | 35 ~RawFile(); |
| 36 | 36 |
| 37 void WriteSamples(const int16_t* samples, size_t num_samples); | 37 void WriteSamples(const int16_t* samples, size_t num_samples); |
| 38 void WriteSamples(const float* samples, size_t num_samples); | 38 void WriteSamples(const float* samples, size_t num_samples); |
| 39 | 39 |
| 40 private: | 40 private: |
| 41 FILE* file_handle_; | 41 FILE* file_handle_; |
| 42 | 42 |
| 43 RTC_DISALLOW_COPY_AND_ASSIGN(RawFile); | 43 RTC_DISALLOW_COPY_AND_ASSIGN(RawFile); |
| 44 }; | 44 }; |
| 45 | 45 |
| 46 // Reads ChannelBuffers from a provided WavReader. | 46 // Reads ChannelBuffers from a provided WavReader. |
| 47 class ChannelBufferWavReader final { | 47 class ChannelBufferWavReader final { |
| 48 public: | 48 public: |
| 49 explicit ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file); | 49 explicit ChannelBufferWavReader(std::unique_ptr<WavReader> file); |
| 50 | 50 |
| 51 // Reads data from the file according to the |buffer| format. Returns false if | 51 // Reads data from the file according to the |buffer| format. Returns false if |
| 52 // a full buffer can't be read from the file. | 52 // a full buffer can't be read from the file. |
| 53 bool Read(ChannelBuffer<float>* buffer); | 53 bool Read(ChannelBuffer<float>* buffer); |
| 54 | 54 |
| 55 private: | 55 private: |
| 56 rtc::scoped_ptr<WavReader> file_; | 56 std::unique_ptr<WavReader> file_; |
| 57 std::vector<float> interleaved_; | 57 std::vector<float> interleaved_; |
| 58 | 58 |
| 59 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader); | 59 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader); |
| 60 }; | 60 }; |
| 61 | 61 |
| 62 // Writes ChannelBuffers to a provided WavWriter. | 62 // Writes ChannelBuffers to a provided WavWriter. |
| 63 class ChannelBufferWavWriter final { | 63 class ChannelBufferWavWriter final { |
| 64 public: | 64 public: |
| 65 explicit ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file); | 65 explicit ChannelBufferWavWriter(std::unique_ptr<WavWriter> file); |
| 66 void Write(const ChannelBuffer<float>& buffer); | 66 void Write(const ChannelBuffer<float>& buffer); |
| 67 | 67 |
| 68 private: | 68 private: |
| 69 rtc::scoped_ptr<WavWriter> file_; | 69 std::unique_ptr<WavWriter> file_; |
| 70 std::vector<float> interleaved_; | 70 std::vector<float> interleaved_; |
| 71 | 71 |
| 72 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter); | 72 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter); |
| 73 }; | 73 }; |
| 74 | 74 |
| 75 void WriteIntData(const int16_t* data, | 75 void WriteIntData(const int16_t* data, |
| 76 size_t length, | 76 size_t length, |
| 77 WavWriter* wav_file, | 77 WavWriter* wav_file, |
| 78 RawFile* raw_file); | 78 RawFile* raw_file); |
| 79 | 79 |
| 80 void WriteFloatData(const float* const* data, | 80 void WriteFloatData(const float* const* data, |
| 81 size_t samples_per_channel, | 81 size_t samples_per_channel, |
| 82 size_t num_channels, | 82 size_t num_channels, |
| 83 WavWriter* wav_file, | 83 WavWriter* wav_file, |
| 84 RawFile* raw_file); | 84 RawFile* raw_file); |
| 85 | 85 |
| 86 // Exits on failure; do not use in unit tests. | 86 // Exits on failure; do not use in unit tests. |
| 87 FILE* OpenFile(const std::string& filename, const char* mode); | 87 FILE* OpenFile(const std::string& filename, const char* mode); |
| 88 | 88 |
| 89 size_t SamplesFromRate(int rate); | 89 size_t SamplesFromRate(int rate); |
| 90 | 90 |
| 91 void SetFrameSampleRate(AudioFrame* frame, | 91 void SetFrameSampleRate(AudioFrame* frame, |
| 92 int sample_rate_hz); | 92 int sample_rate_hz); |
| 93 | 93 |
| 94 template <typename T> | 94 template <typename T> |
| 95 void SetContainerFormat(int sample_rate_hz, | 95 void SetContainerFormat(int sample_rate_hz, |
| 96 size_t num_channels, | 96 size_t num_channels, |
| 97 AudioFrame* frame, | 97 AudioFrame* frame, |
| 98 rtc::scoped_ptr<ChannelBuffer<T> >* cb) { | 98 std::unique_ptr<ChannelBuffer<T> >* cb) { |
| 99 SetFrameSampleRate(frame, sample_rate_hz); | 99 SetFrameSampleRate(frame, sample_rate_hz); |
| 100 frame->num_channels_ = num_channels; | 100 frame->num_channels_ = num_channels; |
| 101 cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels)); | 101 cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels)); |
| 102 } | 102 } |
| 103 | 103 |
| 104 AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels); | 104 AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels); |
| 105 | 105 |
| 106 template <typename T> | 106 template <typename T> |
| 107 float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) { | 107 float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) { |
| 108 float mse = 0; | 108 float mse = 0; |
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| 146 // appropriate error message has been printed to stdout. | 146 // appropriate error message has been printed to stdout. |
| 147 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions, | 147 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions, |
| 148 size_t num_mics); | 148 size_t num_mics); |
| 149 | 149 |
| 150 // Same as above, but without the num_mics check for when it isn't available. | 150 // Same as above, but without the num_mics check for when it isn't available. |
| 151 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions); | 151 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions); |
| 152 | 152 |
| 153 } // namespace webrtc | 153 } // namespace webrtc |
| 154 | 154 |
| 155 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ | 155 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ |
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