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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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 |
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34 ~AecDumpBasedSimulator() override; | 34 ~AecDumpBasedSimulator() override; |
35 | 35 |
36 // Processes the messages in the aecdump file. | 36 // Processes the messages in the aecdump file. |
37 void Process() override; | 37 void Process() override; |
38 | 38 |
39 private: | 39 private: |
40 void HandleMessage(const webrtc::audioproc::Init& msg); | 40 void HandleMessage(const webrtc::audioproc::Init& msg); |
41 void HandleMessage(const webrtc::audioproc::Stream& msg); | 41 void HandleMessage(const webrtc::audioproc::Stream& msg); |
42 void HandleMessage(const webrtc::audioproc::ReverseStream& msg); | 42 void HandleMessage(const webrtc::audioproc::ReverseStream& msg); |
43 void HandleMessage(const webrtc::audioproc::Config& msg); | 43 void HandleMessage(const webrtc::audioproc::Config& msg); |
44 void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg, | 44 void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg); |
45 bool* set_stream_analog_level_called); | |
46 void PrepareReverseProcessStreamCall( | 45 void PrepareReverseProcessStreamCall( |
47 const webrtc::audioproc::ReverseStream& msg); | 46 const webrtc::audioproc::ReverseStream& msg); |
48 void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg); | 47 void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg); |
49 | 48 |
50 enum InterfaceType { | 49 enum InterfaceType { |
51 kFixedInterface, | 50 kFixedInterface, |
52 kFloatInterface, | 51 kFloatInterface, |
53 kNotSpecified, | 52 kNotSpecified, |
54 }; | 53 }; |
55 | 54 |
56 FILE* dump_input_file_; | 55 FILE* dump_input_file_; |
57 std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_; | 56 std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_; |
58 std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_; | 57 std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_; |
59 bool artificial_nearend_eof_reported_ = false; | 58 bool artificial_nearend_eof_reported_ = false; |
60 InterfaceType interface_used_ = InterfaceType::kNotSpecified; | 59 InterfaceType interface_used_ = InterfaceType::kNotSpecified; |
61 | 60 |
62 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator); | 61 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator); |
63 }; | 62 }; |
64 | 63 |
65 } // namespace test | 64 } // namespace test |
66 } // namespace webrtc | 65 } // namespace webrtc |
67 | 66 |
68 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ | 67 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ |
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