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Side by Side Diff: webrtc/modules/audio_processing/test/audio_file_processor.h

Issue 1694423002: Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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_AUDIO_FILE_PROCESSOR_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
13 13
14 #include <algorithm> 14 #include <algorithm>
15 #include <limits> 15 #include <limits>
16 #include <memory>
16 #include <vector> 17 #include <vector>
17 18
18 #include "webrtc/base/scoped_ptr.h"
19 #include "webrtc/common_audio/channel_buffer.h" 19 #include "webrtc/common_audio/channel_buffer.h"
20 #include "webrtc/common_audio/wav_file.h" 20 #include "webrtc/common_audio/wav_file.h"
21 #include "webrtc/modules/audio_processing/include/audio_processing.h" 21 #include "webrtc/modules/audio_processing/include/audio_processing.h"
22 #include "webrtc/modules/audio_processing/test/test_utils.h" 22 #include "webrtc/modules/audio_processing/test/test_utils.h"
23 #include "webrtc/system_wrappers/include/tick_util.h" 23 #include "webrtc/system_wrappers/include/tick_util.h"
24 24
25 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD 25 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
26 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" 26 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
27 #else 27 #else
28 #include "webrtc/modules/audio_processing/debug.pb.h" 28 #include "webrtc/modules/audio_processing/debug.pb.h"
(...skipping 48 matching lines...) Expand 10 before | Expand all | Expand 10 after
77 TickIntervalStats* mutable_proc_time() { return &proc_time_; } 77 TickIntervalStats* mutable_proc_time() { return &proc_time_; }
78 78
79 private: 79 private:
80 TickIntervalStats proc_time_; 80 TickIntervalStats proc_time_;
81 }; 81 };
82 82
83 // Used to read from and write to WavFile objects. 83 // Used to read from and write to WavFile objects.
84 class WavFileProcessor final : public AudioFileProcessor { 84 class WavFileProcessor final : public AudioFileProcessor {
85 public: 85 public:
86 // Takes ownership of all parameters. 86 // Takes ownership of all parameters.
87 WavFileProcessor(rtc::scoped_ptr<AudioProcessing> ap, 87 WavFileProcessor(std::unique_ptr<AudioProcessing> ap,
88 rtc::scoped_ptr<WavReader> in_file, 88 std::unique_ptr<WavReader> in_file,
89 rtc::scoped_ptr<WavWriter> out_file); 89 std::unique_ptr<WavWriter> out_file);
90 virtual ~WavFileProcessor() {} 90 virtual ~WavFileProcessor() {}
91 91
92 // Processes one chunk from the WAV input and writes to the WAV output. 92 // Processes one chunk from the WAV input and writes to the WAV output.
93 bool ProcessChunk() override; 93 bool ProcessChunk() override;
94 94
95 private: 95 private:
96 rtc::scoped_ptr<AudioProcessing> ap_; 96 std::unique_ptr<AudioProcessing> ap_;
97 97
98 ChannelBuffer<float> in_buf_; 98 ChannelBuffer<float> in_buf_;
99 ChannelBuffer<float> out_buf_; 99 ChannelBuffer<float> out_buf_;
100 const StreamConfig input_config_; 100 const StreamConfig input_config_;
101 const StreamConfig output_config_; 101 const StreamConfig output_config_;
102 ChannelBufferWavReader buffer_reader_; 102 ChannelBufferWavReader buffer_reader_;
103 ChannelBufferWavWriter buffer_writer_; 103 ChannelBufferWavWriter buffer_writer_;
104 }; 104 };
105 105
106 // Used to read from an aecdump file and write to a WavWriter. 106 // Used to read from an aecdump file and write to a WavWriter.
107 class AecDumpFileProcessor final : public AudioFileProcessor { 107 class AecDumpFileProcessor final : public AudioFileProcessor {
108 public: 108 public:
109 // Takes ownership of all parameters. 109 // Takes ownership of all parameters.
110 AecDumpFileProcessor(rtc::scoped_ptr<AudioProcessing> ap, 110 AecDumpFileProcessor(std::unique_ptr<AudioProcessing> ap,
111 FILE* dump_file, 111 FILE* dump_file,
112 rtc::scoped_ptr<WavWriter> out_file); 112 std::unique_ptr<WavWriter> out_file);
113 113
114 virtual ~AecDumpFileProcessor(); 114 virtual ~AecDumpFileProcessor();
115 115
116 // Processes messages from the aecdump file until the first Stream message is 116 // Processes messages from the aecdump file until the first Stream message is
117 // completed. Passes other data from the aecdump messages as appropriate. 117 // completed. Passes other data from the aecdump messages as appropriate.
118 bool ProcessChunk() override; 118 bool ProcessChunk() override;
119 119
120 private: 120 private:
121 void HandleMessage(const webrtc::audioproc::Init& msg); 121 void HandleMessage(const webrtc::audioproc::Init& msg);
122 void HandleMessage(const webrtc::audioproc::Stream& msg); 122 void HandleMessage(const webrtc::audioproc::Stream& msg);
123 void HandleMessage(const webrtc::audioproc::ReverseStream& msg); 123 void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
124 124
125 rtc::scoped_ptr<AudioProcessing> ap_; 125 std::unique_ptr<AudioProcessing> ap_;
126 FILE* dump_file_; 126 FILE* dump_file_;
127 127
128 rtc::scoped_ptr<ChannelBuffer<float>> in_buf_; 128 std::unique_ptr<ChannelBuffer<float>> in_buf_;
129 rtc::scoped_ptr<ChannelBuffer<float>> reverse_buf_; 129 std::unique_ptr<ChannelBuffer<float>> reverse_buf_;
130 ChannelBuffer<float> out_buf_; 130 ChannelBuffer<float> out_buf_;
131 StreamConfig input_config_; 131 StreamConfig input_config_;
132 StreamConfig reverse_config_; 132 StreamConfig reverse_config_;
133 const StreamConfig output_config_; 133 const StreamConfig output_config_;
134 ChannelBufferWavWriter buffer_writer_; 134 ChannelBufferWavWriter buffer_writer_;
135 }; 135 };
136 136
137 } // namespace webrtc 137 } // namespace webrtc
138 138
139 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_ 139 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
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