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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_DUMP_CAPTURE_STREAM_INFO_H_ | |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_ | |
13 | |
14 #include <memory> | |
15 #include <utility> | |
16 #include <vector> | |
17 | |
18 #include "webrtc/base/checks.h" | |
19 #include "webrtc/base/ignore_wundef.h" | |
20 #include "webrtc/base/logging.h" | |
21 #include "webrtc/modules/audio_processing/aec_dump/write_to_file_task.h" | |
22 #include "webrtc/modules/audio_processing/include/aec_dump.h" | |
23 #include "webrtc/modules/include/module_common_types.h" | |
24 | |
25 // Files generated at build-time by the protobuf compiler. | |
26 RTC_PUSH_IGNORING_WUNDEF() | |
27 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | |
28 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" | |
29 #else | |
30 #include "webrtc/modules/audio_processing/debug.pb.h" | |
31 #endif | |
32 RTC_POP_IGNORING_WUNDEF() | |
33 | |
34 namespace webrtc { | |
35 | |
36 class CaptureStreamInfo { | |
37 public: | |
38 explicit CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task); | |
39 ~CaptureStreamInfo(); | |
40 void AddInput(const FloatAudioFrame& src); | |
41 void AddOutput(const FloatAudioFrame& src); | |
42 | |
43 void AddInput(const AudioFrame& frame); | |
44 void AddOutput(const AudioFrame& frame); | |
45 | |
46 void AddAudioProcessingState(const AudioProcessingState& state); | |
47 | |
48 std::unique_ptr<WriteToFileTask> GetTask() { | |
49 RTC_DCHECK(task_); | |
50 return std::move(task_); | |
51 } | |
52 | |
53 void SetTask(std::unique_ptr<WriteToFileTask> task) { | |
54 RTC_DCHECK(!task_); | |
peah-webrtc
2017/05/16 06:30:38
You probably should add
RTC_DCHECK(task);
as well
aleloi
2017/05/16 20:10:17
Done.
| |
55 task_ = std::move(task); | |
56 } | |
57 | |
58 private: | |
59 std::unique_ptr<WriteToFileTask> task_; | |
60 }; | |
61 | |
62 } // namespace webrtc | |
63 | |
64 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_ | |
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