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/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_MEDIA_WEBRTCVOE_H_
#define TALK_MEDIA_WEBRTCVOE_H_
#include "talk/media/webrtc/webrtccommon.h"
#include "webrtc/base/common.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_errors.h"
#include "webrtc/voice_engine/include/voe_hardware.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
namespace cricket {
// automatically handles lifetime of WebRtc VoiceEngine
class scoped_voe_engine {
public:
explicit scoped_voe_engine(webrtc::VoiceEngine* e) : ptr(e) {}
// VERIFY, to ensure that there are no leaks at shutdown
~scoped_voe_engine() { if (ptr) VERIFY(webrtc::VoiceEngine::Delete(ptr)); }
// Releases the current pointer.
void reset() {
if (ptr) {
VERIFY(webrtc::VoiceEngine::Delete(ptr));
ptr = NULL;
}
}
webrtc::VoiceEngine* get() const { return ptr; }
private:
webrtc::VoiceEngine* ptr;
};
// scoped_ptr class to handle obtaining and releasing WebRTC interface pointers
template<class T>
class scoped_voe_ptr {
public:
explicit scoped_voe_ptr(const scoped_voe_engine& e)
: ptr(T::GetInterface(e.get())) {}
explicit scoped_voe_ptr(T* p) : ptr(p) {}
~scoped_voe_ptr() { if (ptr) ptr->Release(); }
T* operator->() const { return ptr; }
T* get() const { return ptr; }
// Releases the current pointer.
void reset() {
if (ptr) {
ptr->Release();
ptr = NULL;
}
}
private:
T* ptr;
};
// Utility class for aggregating the various WebRTC interface.
// Fake implementations can also be injected for testing.
class VoEWrapper {
public:
VoEWrapper()
: engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
base_(engine_), codec_(engine_),
hw_(engine_), network_(engine_),
rtp_(engine_), volume_(engine_) {
}
VoEWrapper(webrtc::VoEAudioProcessing* processing,
webrtc::VoEBase* base,
webrtc::VoECodec* codec,
webrtc::VoEHardware* hw,
webrtc::VoENetwork* network,
webrtc::VoERTP_RTCP* rtp,
webrtc::VoEVolumeControl* volume)
: engine_(NULL),
processing_(processing),
base_(base),
codec_(codec),
hw_(hw),
network_(network),
rtp_(rtp),
volume_(volume) {
}
~VoEWrapper() {}
webrtc::VoiceEngine* engine() const { return engine_.get(); }
webrtc::VoEAudioProcessing* processing() const { return processing_.get(); }
webrtc::VoEBase* base() const { return base_.get(); }
webrtc::VoECodec* codec() const { return codec_.get(); }
webrtc::VoEHardware* hw() const { return hw_.get(); }
webrtc::VoENetwork* network() const { return network_.get(); }
webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
int error() { return base_->LastError(); }
private:
scoped_voe_engine engine_;
scoped_voe_ptr<webrtc::VoEAudioProcessing> processing_;
scoped_voe_ptr<webrtc::VoEBase> base_;
scoped_voe_ptr<webrtc::VoECodec> codec_;
scoped_voe_ptr<webrtc::VoEHardware> hw_;
scoped_voe_ptr<webrtc::VoENetwork> network_;
scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
};
} // namespace cricket
#endif // TALK_MEDIA_WEBRTCVOE_H_
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