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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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359 // Only the rate and samples fields of capture_processing_format_ are used | 359 // Only the rate and samples fields of capture_processing_format_ are used |
360 // because the forward processing number of channels is mutable and is | 360 // because the forward processing number of channels is mutable and is |
361 // tracked by the capture_audio_. | 361 // tracked by the capture_audio_. |
362 StreamConfig capture_processing_format; | 362 StreamConfig capture_processing_format; |
363 int split_rate; | 363 int split_rate; |
364 int stream_delay_ms; | 364 int stream_delay_ms; |
365 bool beamformer_enabled; | 365 bool beamformer_enabled; |
366 bool intelligibility_enabled; | 366 bool intelligibility_enabled; |
367 bool level_controller_enabled = false; | 367 bool level_controller_enabled = false; |
368 bool echo_canceller3_enabled = false; | 368 bool echo_canceller3_enabled = false; |
369 bool gain_controller2_enabled = false; | |
370 } capture_nonlocked_; | 369 } capture_nonlocked_; |
371 | 370 |
372 struct ApmRenderState { | 371 struct ApmRenderState { |
373 ApmRenderState(); | 372 ApmRenderState(); |
374 ~ApmRenderState(); | 373 ~ApmRenderState(); |
375 std::unique_ptr<AudioConverter> render_converter; | 374 std::unique_ptr<AudioConverter> render_converter; |
376 std::unique_ptr<AudioBuffer> render_audio; | 375 std::unique_ptr<AudioBuffer> render_audio; |
377 } render_ RTC_GUARDED_BY(crit_render_); | 376 } render_ RTC_GUARDED_BY(crit_render_); |
378 | 377 |
379 size_t aec_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_) | 378 size_t aec_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_) |
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409 std::unique_ptr< | 408 std::unique_ptr< |
410 SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> | 409 SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> |
411 agc_render_signal_queue_; | 410 agc_render_signal_queue_; |
412 std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>> | 411 std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>> |
413 red_render_signal_queue_; | 412 red_render_signal_queue_; |
414 }; | 413 }; |
415 | 414 |
416 } // namespace webrtc | 415 } // namespace webrtc |
417 | 416 |
418 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ | 417 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
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