d4e598d57a
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
73 lines
2.6 KiB
C++
73 lines
2.6 KiB
C++
/*
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* libjingle
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* Copyright 2014, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "talk/app/webrtc/remoteaudiosource.h"
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#include <algorithm>
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#include <functional>
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#include "webrtc/base/logging.h"
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namespace webrtc {
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rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create() {
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return new rtc::RefCountedObject<RemoteAudioSource>();
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}
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RemoteAudioSource::RemoteAudioSource() {
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}
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RemoteAudioSource::~RemoteAudioSource() {
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ASSERT(audio_observers_.empty());
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}
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MediaSourceInterface::SourceState RemoteAudioSource::state() const {
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return MediaSourceInterface::kLive;
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}
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void RemoteAudioSource::SetVolume(double volume) {
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ASSERT(volume >= 0 && volume <= 10);
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for (AudioObserverList::iterator it = audio_observers_.begin();
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it != audio_observers_.end(); ++it) {
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(*it)->OnSetVolume(volume);
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}
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}
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void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
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ASSERT(observer != NULL);
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ASSERT(std::find(audio_observers_.begin(), audio_observers_.end(),
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observer) == audio_observers_.end());
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audio_observers_.push_back(observer);
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}
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void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
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ASSERT(observer != NULL);
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audio_observers_.remove(observer);
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}
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} // namespace webrtc
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