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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
42 lines
1.3 KiB
C++
42 lines
1.3 KiB
C++
/*
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* Copyright 2016 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_DEVICE_IOS_AUDIO_SESSION_OBSERVER_H_
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#define MODULES_AUDIO_DEVICE_IOS_AUDIO_SESSION_OBSERVER_H_
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#include "rtc_base/asyncinvoker.h"
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#include "rtc_base/thread.h"
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namespace webrtc {
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// Observer interface for listening to AVAudioSession events.
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class AudioSessionObserver {
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public:
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// Called when audio session interruption begins.
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virtual void OnInterruptionBegin() = 0;
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// Called when audio session interruption ends.
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virtual void OnInterruptionEnd() = 0;
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// Called when audio route changes.
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virtual void OnValidRouteChange() = 0;
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// Called when the ability to play or record changes.
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virtual void OnCanPlayOrRecordChange(bool can_play_or_record) = 0;
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virtual void OnChangedOutputVolume() = 0;
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protected:
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virtual ~AudioSessionObserver() {}
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_IOS_AUDIO_SESSION_OBSERVER_H_
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