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

Issue 1678813002: Moved the GainControlForNewAGC class to a separate file. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Changed the locking test to match how the gain control is used in practice and to comply with the n… Created 4 years, 10 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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267 {kSampleRate16kHz, 1, false}, 267 {kSampleRate16kHz, 1, false},
268 {kSampleRate16kHz, 1, false}}}), 268 {kSampleRate16kHz, 1, false}}}),
269 rev_proc_format(kSampleRate16kHz, 1) {} 269 rev_proc_format(kSampleRate16kHz, 1) {}
270 ProcessingConfig api_format; 270 ProcessingConfig api_format;
271 StreamConfig rev_proc_format; 271 StreamConfig rev_proc_format;
272 } formats_; 272 } formats_;
273 273
274 // APM constants. 274 // APM constants.
275 const struct ApmConstants { 275 const struct ApmConstants {
276 ApmConstants(int agc_startup_min_volume, 276 ApmConstants(int agc_startup_min_volume,
277 bool use_new_agc, 277 bool use_experimental_agc,
278 bool intelligibility_enabled) 278 bool intelligibility_enabled)
279 : // Format of processing streams at input/output call sites. 279 : // Format of processing streams at input/output call sites.
280 agc_startup_min_volume(agc_startup_min_volume), 280 agc_startup_min_volume(agc_startup_min_volume),
281 use_new_agc(use_new_agc), 281 use_experimental_agc(use_experimental_agc),
282 intelligibility_enabled(intelligibility_enabled) {} 282 intelligibility_enabled(intelligibility_enabled) {}
283 int agc_startup_min_volume; 283 int agc_startup_min_volume;
284 bool use_new_agc; 284 bool use_experimental_agc;
285 bool intelligibility_enabled; 285 bool intelligibility_enabled;
286 } constants_; 286 } constants_;
287 287
288 struct ApmCaptureState { 288 struct ApmCaptureState {
289 ApmCaptureState(bool transient_suppressor_enabled, 289 ApmCaptureState(bool transient_suppressor_enabled,
290 const std::vector<Point>& array_geometry, 290 const std::vector<Point>& array_geometry,
291 SphericalPointf target_direction) 291 SphericalPointf target_direction)
292 : aec_system_delay_jumps(-1), 292 : aec_system_delay_jumps(-1),
293 delay_offset_ms(0), 293 delay_offset_ms(0),
294 was_stream_delay_set(false), 294 was_stream_delay_set(false),
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338 338
339 struct ApmRenderState { 339 struct ApmRenderState {
340 rtc::scoped_ptr<AudioConverter> render_converter; 340 rtc::scoped_ptr<AudioConverter> render_converter;
341 rtc::scoped_ptr<AudioBuffer> render_audio; 341 rtc::scoped_ptr<AudioBuffer> render_audio;
342 } render_ GUARDED_BY(crit_render_); 342 } render_ GUARDED_BY(crit_render_);
343 }; 343 };
344 344
345 } // namespace webrtc 345 } // namespace webrtc
346 346
347 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ 347 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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