webrtc/third_party_mods/libjingle/source/talk/session/phone/webrtcvie.h
ronghuawu@google.com e256187f8b * Push the //depotGoogle/chrome/third_party/libjingle/...@38654 to svn third_party_mods\libjingle.
* Update the peerconnection sample client accordingly.
Review URL: http://webrtc-codereview.appspot.com/60008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@302 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-04 17:44:30 +00:00

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5.5 KiB
C++

/*
* libjingle
* Copyright 2004--2011, 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_SESSION_PHONE_WEBRTCVIE_H_
#define TALK_SESSION_PHONE_WEBRTCVIE_H_
#include "talk/base/common.h"
#include "talk/session/phone/webrtccommon.h"
#ifdef WEBRTC_RELATIVE_PATH
#include "common_types.h"
#include "modules/interface/module_common_types.h"
#include "modules/video_capture/main/interface/video_capture.h"
#include "modules/video_render/main/interface/video_render.h"
#include "video_engine/main/interface/vie_base.h"
#include "video_engine/main/interface/vie_capture.h"
#include "video_engine/main/interface/vie_codec.h"
#include "video_engine/main/interface/vie_errors.h"
#include "video_engine/main/interface/vie_image_process.h"
#include "video_engine/main/interface/vie_network.h"
#include "video_engine/main/interface/vie_render.h"
#include "video_engine/main/interface/vie_rtp_rtcp.h"
#else
#include "third_party/webrtc/files/include/common_types.h"
#include "third_party/webrtc/files/include/module_common_types.h"
#include "third_party/webrtc/files/include/video_capture.h"
#include "third_party/webrtc/files/include/video_render.h"
#include "third_party/webrtc/files/include/vie_base.h"
#include "third_party/webrtc/files/include/vie_capture.h"
#include "third_party/webrtc/files/include/vie_codec.h"
#include "third_party/webrtc/files/include/vie_errors.h"
#include "third_party/webrtc/files/include/vie_image_process.h"
#include "third_party/webrtc/files/include/vie_network.h"
#include "third_party/webrtc/files/include/vie_render.h"
#include "third_party/webrtc/files/include/vie_rtp_rtcp.h"
#endif // WEBRTC_RELATIVE_PATH
namespace cricket {
// all tracing macros should go to a common file
// automatically handles lifetime of VideoEngine
class scoped_vie_engine {
public:
explicit scoped_vie_engine(webrtc::VideoEngine* e) : ptr(e) {}
// VERIFY, to ensure that there are no leaks at shutdown
~scoped_vie_engine() {
if (ptr) {
webrtc::VideoEngine::Delete(ptr);
}
}
webrtc::VideoEngine* get() const { return ptr; }
private:
webrtc::VideoEngine* ptr;
};
// scoped_ptr class to handle obtaining and releasing VideoEngine
// interface pointers
template<class T> class scoped_vie_ptr {
public:
explicit scoped_vie_ptr(const scoped_vie_engine& e)
: ptr(T::GetInterface(e.get())) {}
explicit scoped_vie_ptr(T* p) : ptr(p) {}
~scoped_vie_ptr() { if (ptr) ptr->Release(); }
T* operator->() const { return ptr; }
T* get() const { return ptr; }
private:
T* ptr;
};
// Utility class for aggregating the various WebRTC interface.
// Fake implementations can also be injected for testing.
class ViEWrapper {
public:
ViEWrapper()
: engine_(webrtc::VideoEngine::Create()),
base_(engine_), codec_(engine_), capture_(engine_),
network_(engine_), render_(engine_), rtp_(engine_),
image_(engine_) {
}
ViEWrapper(webrtc::ViEBase* base, webrtc::ViECodec* codec,
webrtc::ViECapture* capture, webrtc::ViENetwork* network,
webrtc::ViERender* render, webrtc::ViERTP_RTCP* rtp,
webrtc::ViEImageProcess* image)
: engine_(NULL),
base_(base),
codec_(codec),
capture_(capture),
network_(network),
render_(render),
rtp_(rtp),
image_(image) {
}
virtual ~ViEWrapper() {}
webrtc::VideoEngine* engine() { return engine_.get(); }
webrtc::ViEBase* base() { return base_.get(); }
webrtc::ViECodec* codec() { return codec_.get(); }
webrtc::ViECapture* capture() { return capture_.get(); }
webrtc::ViENetwork* network() { return network_.get(); }
webrtc::ViERender* render() { return render_.get(); }
webrtc::ViERTP_RTCP* rtp() { return rtp_.get(); }
webrtc::ViEImageProcess* sync() { return image_.get(); }
int error() { return base_->LastError(); }
private:
scoped_vie_engine engine_;
scoped_vie_ptr<webrtc::ViEBase> base_;
scoped_vie_ptr<webrtc::ViECodec> codec_;
scoped_vie_ptr<webrtc::ViECapture> capture_;
scoped_vie_ptr<webrtc::ViENetwork> network_;
scoped_vie_ptr<webrtc::ViERender> render_;
scoped_vie_ptr<webrtc::ViERTP_RTCP> rtp_;
scoped_vie_ptr<webrtc::ViEImageProcess> image_;
};
}
#endif // TALK_SESSION_PHONE_WEBRTCVIE_H_