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1 /* | |
2 * Copyright (c) 2017 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_AEC_DUMPER_AEC_DUMPER_H_ | |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMPER_AEC_DUMPER_H_ | |
13 | |
14 #include <memory> | |
15 #include <string> | |
16 #include <vector> | |
17 | |
18 #include "webrtc/base/array_view.h" | |
19 #include "webrtc/modules/audio_processing/include/audio_processing.h" | |
20 | |
21 namespace audioproc { | |
22 class Event; | |
23 } // namespace audioproc | |
24 | |
25 namespace rtc { | |
26 class TaskQueue; | |
27 } // namespace rtc | |
28 | |
29 namespace webrtc { | |
30 | |
31 class AudioFrame; | |
32 | |
33 // Struct for passing current config from APM without having to | |
34 // include protobuf headers. | |
35 struct InternalAPMConfig { | |
36 InternalAPMConfig(); | |
37 | |
38 bool aec_enabled = false; | |
39 bool aec_delay_agnostic_enabled = false; | |
40 bool aec_drift_compensation_enabled = false; | |
41 bool aec_extended_filter_enabled = false; | |
42 int aec_suppression_level = 0; | |
43 bool aecm_enabled = false; | |
44 bool aecm_comfort_noise_enabled = false; | |
45 int aecm_routing_mode = 0; | |
46 bool agc_enabled = false; | |
47 int agc_mode = 0; | |
48 bool agc_limiter_enabled = false; | |
49 bool hpf_enabled = false; | |
50 bool ns_enabled = false; | |
51 int ns_level = 0; | |
52 bool transient_suppression_enabled = false; | |
53 bool intelligibility_enhancer_enabled = false; | |
54 bool noise_robust_agc_enabled = false; | |
55 std::string experiments_description = ""; | |
56 | |
57 private: | |
58 RTC_DISALLOW_COPY_AND_ASSIGN(InternalAPMConfig); | |
kwiberg-webrtc
2017/03/29 08:57:11
You don't have to use this macro anymore. Just do
| |
59 }; | |
60 | |
61 class AecDumper { | |
62 public: | |
63 // A capture stream frame is logged before and after processing in | |
64 // the same protobuf message. To facilitate that, a | |
65 // CaptureStreamInfo instance is first filled with Input, then | |
66 // Output. | |
67 // | |
68 // To log an input/output pair, first call | |
69 // AecDumper::GetCaptureStreamInfo. Add the input and output to | |
70 // it. Then call AecDumper::WriteCaptureStreamMessage. | |
71 class CaptureStreamInfo { | |
72 public: | |
73 virtual ~CaptureStreamInfo() = default; | |
74 virtual void AddInput(std::vector<rtc::ArrayView<const float>> src) = 0; | |
75 virtual void AddOutput(std::vector<rtc::ArrayView<const float>> src) = 0; | |
76 | |
77 virtual void AddInput(const AudioFrame& frame) = 0; | |
78 virtual void AddOutput(const AudioFrame& frame) = 0; | |
79 | |
80 virtual void set_delay(int delay) = 0; | |
81 virtual void set_drift(int drift) = 0; | |
82 virtual void set_level(int level) = 0; | |
83 virtual void set_keypress(bool keypress) = 0; | |
84 }; | |
85 | |
86 AecDumper() = default; | |
kwiberg-webrtc
2017/03/29 08:57:11
You forgot to remove this one.
| |
87 | |
88 virtual ~AecDumper() = default; | |
89 | |
90 // TODO(aleloi): update comments to new creation scheme. | |
91 // If called when a recording is active, that file is closed, and a | |
92 // new file is opened. Messages waiting to be written asynchronously | |
93 // to the old file may be lost. Returns true iff opening file for | |
94 // writing succeeded. | |
95 | |
96 // Closes associated file. Messages waiting to be written to file | |
97 // asynchronously may be lost. This method is safe to call when no | |
98 // recording is active. A recording does not have to be closed | |
99 // manually with this method; instead the AecDumper instance may be | |
100 // destroyed. | |
101 static std::unique_ptr<AecDumper> Create(std::string file_name, | |
102 int64_t max_log_size_bytes, | |
103 rtc::TaskQueue* worker_queue); | |
104 static std::unique_ptr<AecDumper> Create(FILE* handle, | |
105 int64_t max_log_size_bytes, | |
106 rtc::TaskQueue* worker_queue); | |
107 | |
108 static std::unique_ptr<AecDumper> CreateNullDumper(); | |
109 | |
110 virtual std::unique_ptr<CaptureStreamInfo> GetCaptureStreamInfo() = 0; | |
111 | |
112 // The Write* methods are always safe to call. If no recording is in | |
113 // progress, calls will have no effect. Messages are written to file | |
114 // in a 'best effort' manner. If the AecDumper can't keep up with | |
115 // the flow of messages, some will be silently dropped. | |
116 virtual void WriteInitMessage(const ProcessingConfig& api_format) = 0; | |
117 | |
118 virtual void WriteReverseStreamMessage(const AudioFrame& frame) = 0; | |
119 | |
120 virtual void WriteReverseStreamMessage( | |
121 std::vector<rtc::ArrayView<const float>> src) = 0; | |
122 | |
123 virtual void WriteCaptureStreamMessage( | |
124 std::unique_ptr<CaptureStreamInfo> stream_info) = 0; | |
125 | |
126 // If not |forced|, only writes the current config if it is | |
127 // different from the last saved one; if |forced|, writes the config | |
128 // regardless of the last saved. | |
129 virtual void WriteConfig(const InternalAPMConfig& config, bool forced) = 0; | |
130 }; | |
131 } // namespace webrtc | |
132 | |
133 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMPER_AEC_DUMPER_H_ | |
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