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

Issue 2822253002: Minor correction of AudioProcessing::set_stream_delay_ms() docstring (Closed)
Patch Set: Created 3 years, 8 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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422 // 422 //
423 // Sets the |delay| in ms between ProcessReverseStream() receiving a far-end 423 // Sets the |delay| in ms between ProcessReverseStream() receiving a far-end
424 // frame and ProcessStream() receiving a near-end frame containing the 424 // frame and ProcessStream() receiving a near-end frame containing the
425 // corresponding echo. On the client-side this can be expressed as 425 // corresponding echo. On the client-side this can be expressed as
426 // delay = (t_render - t_analyze) + (t_process - t_capture) 426 // delay = (t_render - t_analyze) + (t_process - t_capture)
427 // where, 427 // where,
428 // - t_analyze is the time a frame is passed to ProcessReverseStream() and 428 // - t_analyze is the time a frame is passed to ProcessReverseStream() and
429 // t_render is the time the first sample of the same frame is rendered by 429 // t_render is the time the first sample of the same frame is rendered by
430 // the audio hardware. 430 // the audio hardware.
431 // - t_capture is the time the first sample of a frame is captured by the 431 // - t_capture is the time the first sample of a frame is captured by the
432 // audio hardware and t_pull is the time the same frame is passed to 432 // audio hardware and t_process is the time the same frame is passed to
433 // ProcessStream(). 433 // ProcessStream().
434 virtual int set_stream_delay_ms(int delay) = 0; 434 virtual int set_stream_delay_ms(int delay) = 0;
435 virtual int stream_delay_ms() const = 0; 435 virtual int stream_delay_ms() const = 0;
436 virtual bool was_stream_delay_set() const = 0; 436 virtual bool was_stream_delay_set() const = 0;
437 437
438 // Call to signal that a key press occurred (true) or did not occur (false) 438 // Call to signal that a key press occurred (true) or did not occur (false)
439 // with this chunk of audio. 439 // with this chunk of audio.
440 virtual void set_stream_key_pressed(bool key_pressed) = 0; 440 virtual void set_stream_key_pressed(bool key_pressed) = 0;
441 441
442 // Sets a delay |offset| in ms to add to the values passed in through 442 // Sets a delay |offset| in ms to add to the values passed in through
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1074 // This does not impact the size of frames passed to |ProcessStream()|. 1074 // This does not impact the size of frames passed to |ProcessStream()|.
1075 virtual int set_frame_size_ms(int size) = 0; 1075 virtual int set_frame_size_ms(int size) = 0;
1076 virtual int frame_size_ms() const = 0; 1076 virtual int frame_size_ms() const = 0;
1077 1077
1078 protected: 1078 protected:
1079 virtual ~VoiceDetection() {} 1079 virtual ~VoiceDetection() {}
1080 }; 1080 };
1081 } // namespace webrtc 1081 } // namespace webrtc
1082 1082
1083 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ 1083 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
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