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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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573 RTCLog(@"Handling playout sample rate change to: %f\n" | 573 RTCLog(@"Handling playout sample rate change to: %f\n" |
574 " Session sample rate: %f frames_per_buffer: %lu\n" | 574 " Session sample rate: %f frames_per_buffer: %lu\n" |
575 " ADM sample rate: %f frames_per_buffer: %lu", | 575 " ADM sample rate: %f frames_per_buffer: %lu", |
576 sample_rate, | 576 sample_rate, |
577 session_sample_rate, | 577 session_sample_rate, |
578 (unsigned long)session_frames_per_buffer, | 578 (unsigned long)session_frames_per_buffer, |
579 current_sample_rate, | 579 current_sample_rate, |
580 (unsigned long)current_frames_per_buffer); | 580 (unsigned long)current_frames_per_buffer); |
581 | 581 |
582 // Sample rate and buffer size are the same, no work to do. | 582 // Sample rate and buffer size are the same, no work to do. |
583 if (std::abs(current_sample_rate - session_sample_rate) <= DBL_EPSILON && | 583 if (std::abs((int)current_sample_rate - (int)session_sample_rate) <= DBL_EPSIL
ON && |
584 current_frames_per_buffer == session_frames_per_buffer) { | 584 current_frames_per_buffer == session_frames_per_buffer) { |
585 return; | 585 return; |
586 } | 586 } |
587 | 587 |
588 // We need to adjust our format and buffer sizes. | 588 // We need to adjust our format and buffer sizes. |
589 // The stream format is about to be changed and it requires that we first | 589 // The stream format is about to be changed and it requires that we first |
590 // stop and uninitialize the audio unit to deallocate its resources. | 590 // stop and uninitialize the audio unit to deallocate its resources. |
591 RTCLog(@"Stopping and uninitializing audio unit to adjust buffers."); | 591 RTCLog(@"Stopping and uninitializing audio unit to adjust buffers."); |
592 bool restart_audio_unit = false; | 592 bool restart_audio_unit = false; |
593 if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) { | 593 if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) { |
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869 | 869 |
870 // All I/O should be stopped or paused prior to deactivating the audio | 870 // All I/O should be stopped or paused prior to deactivating the audio |
871 // session, hence we deactivate as last action. | 871 // session, hence we deactivate as last action. |
872 [session lockForConfiguration]; | 872 [session lockForConfiguration]; |
873 UnconfigureAudioSession(); | 873 UnconfigureAudioSession(); |
874 [session endWebRTCSession:nil]; | 874 [session endWebRTCSession:nil]; |
875 [session unlockForConfiguration]; | 875 [session unlockForConfiguration]; |
876 } | 876 } |
877 | 877 |
878 } // namespace webrtc | 878 } // namespace webrtc |
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