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Side by Side Diff: webrtc/modules/audio_processing/include/audio_processing.h

Issue 2778783002: AecDump interface (Closed)
Patch Set: Comment formatting. Created 3 years, 7 months ago
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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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448 virtual void set_stream_key_pressed(bool key_pressed) = 0; 449 virtual void set_stream_key_pressed(bool key_pressed) = 0;
449 450
450 // Sets a delay |offset| in ms to add to the values passed in through 451 // Sets a delay |offset| in ms to add to the values passed in through
451 // set_stream_delay_ms(). May be positive or negative. 452 // set_stream_delay_ms(). May be positive or negative.
452 // 453 //
453 // Note that this could cause an otherwise valid value passed to 454 // Note that this could cause an otherwise valid value passed to
454 // set_stream_delay_ms() to return an error. 455 // set_stream_delay_ms() to return an error.
455 virtual void set_delay_offset_ms(int offset) = 0; 456 virtual void set_delay_offset_ms(int offset) = 0;
456 virtual int delay_offset_ms() const = 0; 457 virtual int delay_offset_ms() const = 0;
457 458
459 // Attaches provided webrtc::AecDump for recording debugging
460 // information. Log file and maximum file size logic is supposed to
461 // be handled by implementing instance of AecDump. Calling this
462 // method when another AecDump is attached resets the active AecDump
463 // with a new one. This causes the d-tor of the earlier AecDump to
464 // be called. The d-tor call may block until all pending logging
465 // 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 AttachAecDump(std::unique_ptr<AecDump> aec_dump);
470
471 // If no AecDump is attached, this has no effect. If an AecDump is
472 // attached, it's destructor is called. The d-tor may block until
473 // all pending logging tasks are completed.
474 //
475 // TODO(aleloi): make pure virtual when internal projects have
476 // updated. See https://bugs.webrtc.org/7404
477 virtual void DetachAecDump();
478
458 // Starts recording debugging information to a file specified by |filename|, 479 // Starts recording debugging information to a file specified by |filename|,
459 // a NULL-terminated string. If there is an ongoing recording, the old file 480 // a NULL-terminated string. If there is an ongoing recording, the old file
460 // will be closed, and recording will continue in the newly specified file. 481 // will be closed, and recording will continue in the newly specified file.
461 // An already existing file will be overwritten without warning. A maximum 482 // An already existing file will be overwritten without warning. A maximum
462 // file size (in bytes) for the log can be specified. The logging is stopped 483 // file size (in bytes) for the log can be specified. The logging is stopped
463 // once the limit has been reached. If max_log_size_bytes is set to a value 484 // once the limit has been reached. If max_log_size_bytes is set to a value
464 // <= 0, no limit will be used. 485 // <= 0, no limit will be used.
465 static const size_t kMaxFilenameSize = 1024; 486 static const size_t kMaxFilenameSize = 1024;
466 virtual int StartDebugRecording(const char filename[kMaxFilenameSize], 487 virtual int StartDebugRecording(const char filename[kMaxFilenameSize],
467 int64_t max_log_size_bytes) = 0; 488 int64_t max_log_size_bytes) = 0;
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1082 // This does not impact the size of frames passed to |ProcessStream()|. 1103 // This does not impact the size of frames passed to |ProcessStream()|.
1083 virtual int set_frame_size_ms(int size) = 0; 1104 virtual int set_frame_size_ms(int size) = 0;
1084 virtual int frame_size_ms() const = 0; 1105 virtual int frame_size_ms() const = 0;
1085 1106
1086 protected: 1107 protected:
1087 virtual ~VoiceDetection() {} 1108 virtual ~VoiceDetection() {}
1088 }; 1109 };
1089 } // namespace webrtc 1110 } // namespace webrtc
1090 1111
1091 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ 1112 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
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