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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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484 | 484 |
485 is_interrupted_ = false; | 485 is_interrupted_ = false; |
486 RTCLog(@"Interruption ended. Updating audio unit state."); | 486 RTCLog(@"Interruption ended. Updating audio unit state."); |
487 UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord); | 487 UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord); |
488 } | 488 } |
489 | 489 |
490 void AudioDeviceIOS::HandleValidRouteChange() { | 490 void AudioDeviceIOS::HandleValidRouteChange() { |
491 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 491 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
492 | 492 |
493 RTCAudioSession* session = [RTCAudioSession sharedInstance]; | 493 RTCAudioSession* session = [RTCAudioSession sharedInstance]; |
| 494 RTCLog(@"%@", session); |
494 HandleSampleRateChange(session.sampleRate); | 495 HandleSampleRateChange(session.sampleRate); |
495 } | 496 } |
496 | 497 |
497 void AudioDeviceIOS::HandleCanPlayOrRecordChange(bool can_play_or_record) { | 498 void AudioDeviceIOS::HandleCanPlayOrRecordChange(bool can_play_or_record) { |
498 RTCLog(@"Handling CanPlayOrRecord change to: %d", can_play_or_record); | 499 RTCLog(@"Handling CanPlayOrRecord change to: %d", can_play_or_record); |
499 UpdateAudioUnit(can_play_or_record); | 500 UpdateAudioUnit(can_play_or_record); |
500 } | 501 } |
501 | 502 |
502 void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) { | 503 void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) { |
503 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 504 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
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832 | 833 |
833 // All I/O should be stopped or paused prior to deactivating the audio | 834 // All I/O should be stopped or paused prior to deactivating the audio |
834 // session, hence we deactivate as last action. | 835 // session, hence we deactivate as last action. |
835 [session lockForConfiguration]; | 836 [session lockForConfiguration]; |
836 UnconfigureAudioSession(); | 837 UnconfigureAudioSession(); |
837 [session endWebRTCSession:nil]; | 838 [session endWebRTCSession:nil]; |
838 [session unlockForConfiguration]; | 839 [session unlockForConfiguration]; |
839 } | 840 } |
840 | 841 |
841 } // namespace webrtc | 842 } // namespace webrtc |
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