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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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382 // set_stream_delay_ms(). May be positive or negative. | 382 // set_stream_delay_ms(). May be positive or negative. |
383 // | 383 // |
384 // Note that this could cause an otherwise valid value passed to | 384 // Note that this could cause an otherwise valid value passed to |
385 // set_stream_delay_ms() to return an error. | 385 // set_stream_delay_ms() to return an error. |
386 virtual void set_delay_offset_ms(int offset) = 0; | 386 virtual void set_delay_offset_ms(int offset) = 0; |
387 virtual int delay_offset_ms() const = 0; | 387 virtual int delay_offset_ms() const = 0; |
388 | 388 |
389 // Starts recording debugging information to a file specified by |filename|, | 389 // Starts recording debugging information to a file specified by |filename|, |
390 // a NULL-terminated string. If there is an ongoing recording, the old file | 390 // a NULL-terminated string. If there is an ongoing recording, the old file |
391 // will be closed, and recording will continue in the newly specified file. | 391 // will be closed, and recording will continue in the newly specified file. |
392 // An already existing file will be overwritten without warning. | 392 // An already existing file will be overwritten without warning. A maximum |
393 // file size (in bytes) for the log can be specified. The logging is stopped | |
394 // once the limit has been reached. If max_log_size_bytes is set to a value | |
395 // <= 0, no limit will be used. | |
393 static const size_t kMaxFilenameSize = 1024; | 396 static const size_t kMaxFilenameSize = 1024; |
394 virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0; | 397 virtual int StartDebugRecording(const char filename[kMaxFilenameSize], |
398 int64_t max_log_size_bytes) = 0; | |
395 | 399 |
396 // Same as above but uses an existing file handle. Takes ownership | 400 // Same as above but uses an existing file handle. Takes ownership |
397 // of |handle| and closes it at StopDebugRecording(). | 401 // of |handle| and closes it at StopDebugRecording(). A maximum file size |
398 virtual int StartDebugRecording(FILE* handle) = 0; | 402 // (in bytes) for the log can be specified. The logging is stopped once the |
403 // limit has been reached. | |
Andrew MacDonald
2015/11/24 17:28:59
nit: Just refer to the comments above rather than
ivoc
2015/12/01 15:17:16
Done.
| |
404 virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0; | |
399 | 405 |
400 // Same as above but uses an existing PlatformFile handle. Takes ownership | 406 // Same as above but uses an existing PlatformFile handle. Takes ownership |
401 // of |handle| and closes it at StopDebugRecording(). | 407 // of |handle| and closes it at StopDebugRecording(). |
402 // TODO(xians): Make this interface pure virtual. | 408 // TODO(xians): Make this interface pure virtual. |
403 virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) { | 409 virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) { |
Andrew MacDonald
2015/11/24 17:28:59
Any reason this one doesn't take a max size parame
ivoc
2015/12/01 15:17:16
I think this is the function that Chrome calls to
Andrew MacDonald
2015/12/02 00:11:12
OK. Perhaps mention that no logging limit will be
| |
404 return -1; | 410 return -1; |
405 } | 411 } |
406 | 412 |
407 // Stops recording debugging information, and closes the file. Recording | 413 // Stops recording debugging information, and closes the file. Recording |
408 // cannot be resumed in the same file (without overwriting it). | 414 // cannot be resumed in the same file (without overwriting it). |
409 virtual int StopDebugRecording() = 0; | 415 virtual int StopDebugRecording() = 0; |
410 | 416 |
411 // Use to send UMA histograms at end of a call. Note that all histogram | 417 // Use to send UMA histograms at end of a call. Note that all histogram |
412 // specific member variables are reset. | 418 // specific member variables are reset. |
413 virtual void UpdateHistogramsOnCallEnd() = 0; | 419 virtual void UpdateHistogramsOnCallEnd() = 0; |
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929 // This does not impact the size of frames passed to |ProcessStream()|. | 935 // This does not impact the size of frames passed to |ProcessStream()|. |
930 virtual int set_frame_size_ms(int size) = 0; | 936 virtual int set_frame_size_ms(int size) = 0; |
931 virtual int frame_size_ms() const = 0; | 937 virtual int frame_size_ms() const = 0; |
932 | 938 |
933 protected: | 939 protected: |
934 virtual ~VoiceDetection() {} | 940 virtual ~VoiceDetection() {} |
935 }; | 941 }; |
936 } // namespace webrtc | 942 } // namespace webrtc |
937 | 943 |
938 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ | 944 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ |
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