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1 /* | |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | |
3 * | |
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 | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_PROCESSOR_H_ | |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_PROCESSOR_H_ | |
13 | |
14 #include <algorithm> | |
15 #include <limits> | |
16 #include <memory> | |
17 #include <string> | |
18 #include <vector> | |
19 | |
20 #include "webrtc/base/timeutils.h" | |
21 #include "webrtc/base/optional.h" | |
22 #include "webrtc/common_audio/channel_buffer.h" | |
23 #include "webrtc/common_audio/wav_file.h" | |
24 #include "webrtc/modules/audio_processing/include/audio_processing.h" | |
25 #include "webrtc/modules/audio_processing/test/audio_file_processor.h" | |
26 #include "webrtc/modules/audio_processing/test/test_utils.h" | |
27 | |
28 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | |
29 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" | |
30 #else | |
31 #include "webrtc/modules/audio_processing/debug.pb.h" | |
32 #endif | |
33 | |
34 namespace webrtc { | |
35 namespace test { | |
36 | |
37 // Used to read from an aecdump file and write to a WavWriter. | |
38 class AecDumpFileProcessor final : public AudioFileProcessor { | |
39 public: | |
40 AecDumpFileProcessor(std::unique_ptr<AudioProcessing> ap, | |
41 FILE* dump_file, | |
42 std::string out_filename, | |
43 std::string reverse_out_filename, | |
44 rtc::Optional<int> out_sample_rate_hz, | |
45 rtc::Optional<int> out_num_channels, | |
46 rtc::Optional<int> reverse_out_sample_rate_hz, | |
47 rtc::Optional<int> reverse_out_num_channels, | |
48 bool override_config_message); | |
49 | |
50 virtual ~AecDumpFileProcessor(); | |
51 | |
52 // Processes the messages in the aecdump file and returns | |
53 // the number of forward stream chunks processed. | |
54 size_t Process(bool verbose_logging) override; | |
55 | |
56 private: | |
57 void HandleMessage(const webrtc::audioproc::Init& msg); | |
58 void HandleMessage(const webrtc::audioproc::Stream& msg); | |
59 void HandleMessage(const webrtc::audioproc::ReverseStream& msg); | |
60 void HandleMessage(const webrtc::audioproc::Config& msg); | |
61 | |
62 enum InterfaceType { | |
63 kIntInterface, | |
64 kFloatInterface, | |
65 kNotSpecified, | |
66 }; | |
67 | |
68 std::unique_ptr<AudioProcessing> ap_; | |
69 FILE* dump_file_; | |
70 std::string out_filename_; | |
71 std::string reverse_out_filename_; | |
72 rtc::Optional<int> out_sample_rate_hz_; | |
73 rtc::Optional<int> out_num_channels_; | |
74 rtc::Optional<int> reverse_out_sample_rate_hz_; | |
75 rtc::Optional<int> reverse_out_num_channels_; | |
76 bool override_config_message_; | |
77 | |
78 std::unique_ptr<ChannelBuffer<float>> in_buf_; | |
79 std::unique_ptr<ChannelBuffer<float>> reverse_buf_; | |
80 std::unique_ptr<ChannelBuffer<float>> out_buf_; | |
81 std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_; | |
82 std::unique_ptr<WavWriter> out_file_; | |
83 std::unique_ptr<WavWriter> reverse_out_file_; | |
84 StreamConfig input_config_; | |
85 StreamConfig reverse_config_; | |
86 StreamConfig output_config_; | |
87 StreamConfig reverse_output_config_; | |
88 std::unique_ptr<ChannelBufferWavWriter> buffer_writer_; | |
89 std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_; | |
90 AudioFrame far_frame_; | |
91 AudioFrame near_frame_; | |
92 InterfaceType interface_used_ = InterfaceType::kNotSpecified; | |
93 }; | |
94 | |
95 } // namespace test | |
96 } // namespace webrtc | |
97 | |
98 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_PROCESSOR_H_ | |
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