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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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22 #include "webrtc/base/arraysize.h" | 22 #include "webrtc/base/arraysize.h" |
23 #include "webrtc/base/platform_file.h" | 23 #include "webrtc/base/platform_file.h" |
24 #include "webrtc/modules/audio_processing/beamformer/array_util.h" | 24 #include "webrtc/modules/audio_processing/beamformer/array_util.h" |
25 #include "webrtc/modules/audio_processing/include/config.h" | 25 #include "webrtc/modules/audio_processing/include/config.h" |
26 #include "webrtc/typedefs.h" | 26 #include "webrtc/typedefs.h" |
27 | 27 |
28 namespace webrtc { | 28 namespace webrtc { |
29 | 29 |
30 struct AecCore; | 30 struct AecCore; |
31 | 31 |
| 32 class AecDump; |
32 class AudioFrame; | 33 class AudioFrame; |
33 | 34 |
34 class NonlinearBeamformer; | 35 class NonlinearBeamformer; |
35 | 36 |
36 class StreamConfig; | 37 class StreamConfig; |
37 class ProcessingConfig; | 38 class ProcessingConfig; |
38 | 39 |
39 class EchoCancellation; | 40 class EchoCancellation; |
40 class EchoControlMobile; | 41 class EchoControlMobile; |
41 class GainControl; | 42 class GainControl; |
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440 virtual void set_stream_key_pressed(bool key_pressed) = 0; | 441 virtual void set_stream_key_pressed(bool key_pressed) = 0; |
441 | 442 |
442 // Sets a delay |offset| in ms to add to the values passed in through | 443 // Sets a delay |offset| in ms to add to the values passed in through |
443 // set_stream_delay_ms(). May be positive or negative. | 444 // set_stream_delay_ms(). May be positive or negative. |
444 // | 445 // |
445 // Note that this could cause an otherwise valid value passed to | 446 // Note that this could cause an otherwise valid value passed to |
446 // set_stream_delay_ms() to return an error. | 447 // set_stream_delay_ms() to return an error. |
447 virtual void set_delay_offset_ms(int offset) = 0; | 448 virtual void set_delay_offset_ms(int offset) = 0; |
448 virtual int delay_offset_ms() const = 0; | 449 virtual int delay_offset_ms() const = 0; |
449 | 450 |
| 451 // Attaches provided webrtc::AecDump for recording debugging |
| 452 // information. Log file and maximum file size logic is supposed to |
| 453 // be handled by implementing instance of AecDump. Calling this |
| 454 // method when another AecDump is attached resets the active AecDump |
| 455 // with a new one. This causes the d-tor of the earlier AecDump to |
| 456 // be called. The d-tor call may block until all pending logging |
| 457 // tasks are completed. |
| 458 // |
| 459 // TODO(aleloi): make pure virtual when internal projects have |
| 460 // updated. See https://bugs.webrtc.org/7404 |
| 461 virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump); |
| 462 |
| 463 // If no AecDump is attached, this has no effect. If an AecDump is |
| 464 // attached, it's destructor is called. The d-tor may block until |
| 465 // all pending logging tasks are completed. |
| 466 // |
| 467 // TODO(aleloi): make pure virtual when internal projects have |
| 468 // updated. See https://bugs.webrtc.org/7404 |
| 469 virtual void DetachAecDump(); |
| 470 |
450 // Starts recording debugging information to a file specified by |filename|, | 471 // Starts recording debugging information to a file specified by |filename|, |
451 // a NULL-terminated string. If there is an ongoing recording, the old file | 472 // a NULL-terminated string. If there is an ongoing recording, the old file |
452 // will be closed, and recording will continue in the newly specified file. | 473 // will be closed, and recording will continue in the newly specified file. |
453 // An already existing file will be overwritten without warning. A maximum | 474 // An already existing file will be overwritten without warning. A maximum |
454 // file size (in bytes) for the log can be specified. The logging is stopped | 475 // file size (in bytes) for the log can be specified. The logging is stopped |
455 // once the limit has been reached. If max_log_size_bytes is set to a value | 476 // once the limit has been reached. If max_log_size_bytes is set to a value |
456 // <= 0, no limit will be used. | 477 // <= 0, no limit will be used. |
457 static const size_t kMaxFilenameSize = 1024; | 478 static const size_t kMaxFilenameSize = 1024; |
458 virtual int StartDebugRecording(const char filename[kMaxFilenameSize], | 479 virtual int StartDebugRecording(const char filename[kMaxFilenameSize], |
459 int64_t max_log_size_bytes) = 0; | 480 int64_t max_log_size_bytes) = 0; |
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1074 // This does not impact the size of frames passed to |ProcessStream()|. | 1095 // This does not impact the size of frames passed to |ProcessStream()|. |
1075 virtual int set_frame_size_ms(int size) = 0; | 1096 virtual int set_frame_size_ms(int size) = 0; |
1076 virtual int frame_size_ms() const = 0; | 1097 virtual int frame_size_ms() const = 0; |
1077 | 1098 |
1078 protected: | 1099 protected: |
1079 virtual ~VoiceDetection() {} | 1100 virtual ~VoiceDetection() {} |
1080 }; | 1101 }; |
1081 } // namespace webrtc | 1102 } // namespace webrtc |
1082 | 1103 |
1083 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ | 1104 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ |
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