| /* |
| * 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_; |
| }; |
| |
| // Adds indirection to static WebRtc functions, allowing them to be mocked. |
| class ViETraceWrapper { |
| public: |
| virtual ~ViETraceWrapper() {} |
| |
| virtual int SetTraceFilter(const unsigned int filter) { |
| return webrtc::VideoEngine::SetTraceFilter(filter); |
| } |
| virtual int SetTraceFile(const char* fileNameUTF8) { |
| return webrtc::VideoEngine::SetTraceFile(fileNameUTF8); |
| } |
| virtual int SetTraceCallback(webrtc::TraceCallback* callback) { |
| return webrtc::VideoEngine::SetTraceCallback(callback); |
| } |
| }; |
| |
| } // namespace cricket |
| |
| #endif // TALK_SESSION_PHONE_WEBRTCVIE_H_ |