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

Issue 1525173002: Bugfix that fixes the integration issue that cause WebRTC AEC mobile to fail (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years 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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496 } 496 }
497 497
498 if (capture_.transient_suppressor_enabled != 498 if (capture_.transient_suppressor_enabled !=
499 config.Get<ExperimentalNs>().enabled) { 499 config.Get<ExperimentalNs>().enabled) {
500 capture_.transient_suppressor_enabled = 500 capture_.transient_suppressor_enabled =
501 config.Get<ExperimentalNs>().enabled; 501 config.Get<ExperimentalNs>().enabled;
502 InitializeTransient(); 502 InitializeTransient();
503 } 503 }
504 } 504 }
505 505
506 int AudioProcessingImpl::input_sample_rate_hz() const {
507 // Accessed from outside APM, hence a lock is needed.
508 rtc::CritScope cs(&crit_capture_);
509 return formats_.api_format.input_stream().sample_rate_hz();
510 }
511
506 int AudioProcessingImpl::proc_sample_rate_hz() const { 512 int AudioProcessingImpl::proc_sample_rate_hz() const {
507 // Used as callback from submodules, hence locking is not allowed. 513 // Used as callback from submodules, hence locking is not allowed.
508 return capture_nonlocked_.fwd_proc_format.sample_rate_hz(); 514 return capture_nonlocked_.fwd_proc_format.sample_rate_hz();
509 } 515 }
510 516
511 int AudioProcessingImpl::proc_split_sample_rate_hz() const { 517 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
512 // Used as callback from submodules, hence locking is not allowed. 518 // Used as callback from submodules, hence locking is not allowed.
513 return capture_nonlocked_.split_rate; 519 return capture_nonlocked_.split_rate;
514 } 520 }
515 521
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1479 debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG); 1485 debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
1480 debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config); 1486 debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
1481 1487
1482 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 1488 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
1483 &crit_debug_, &debug_dump_.capture)); 1489 &crit_debug_, &debug_dump_.capture));
1484 return kNoError; 1490 return kNoError;
1485 } 1491 }
1486 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP 1492 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
1487 1493
1488 } // namespace webrtc 1494 } // namespace webrtc
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