1 /* 2 * libjingle 3 * Copyright 2004 Google Inc. 4 * 5 * Redistribution and use in source and binary forms, with or without 6 * modification, are permitted provided that the following conditions are met: 7 * 8 * 1. Redistributions of source code must retain the above copyright notice, 9 * this list of conditions and the following disclaimer. 10 * 2. Redistributions in binary form must reproduce the above copyright notice, 11 * this list of conditions and the following disclaimer in the documentation 12 * and/or other materials provided with the distribution. 13 * 3. The name of the author may not be used to endorse or promote products 14 * derived from this software without specific prior written permission. 15 * 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 26 */ 27 28 #ifndef TALK_MEDIA_WEBRTCVOE_H_ 29 #define TALK_MEDIA_WEBRTCVOE_H_ 30 31 #include "talk/media/webrtc/webrtccommon.h" 32 #include "webrtc/base/common.h" 33 34 #include "webrtc/common_types.h" 35 #include "webrtc/modules/audio_device/include/audio_device.h" 36 #include "webrtc/voice_engine/include/voe_audio_processing.h" 37 #include "webrtc/voice_engine/include/voe_base.h" 38 #include "webrtc/voice_engine/include/voe_codec.h" 39 #include "webrtc/voice_engine/include/voe_errors.h" 40 #include "webrtc/voice_engine/include/voe_hardware.h" 41 #include "webrtc/voice_engine/include/voe_network.h" 42 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" 43 #include "webrtc/voice_engine/include/voe_volume_control.h" 44 45 namespace cricket { 46 // automatically handles lifetime of WebRtc VoiceEngine 47 class scoped_voe_engine { 48 public: scoped_voe_engine(webrtc::VoiceEngine * e)49 explicit scoped_voe_engine(webrtc::VoiceEngine* e) : ptr(e) {} 50 // VERIFY, to ensure that there are no leaks at shutdown ~scoped_voe_engine()51 ~scoped_voe_engine() { if (ptr) VERIFY(webrtc::VoiceEngine::Delete(ptr)); } 52 // Releases the current pointer. reset()53 void reset() { 54 if (ptr) { 55 VERIFY(webrtc::VoiceEngine::Delete(ptr)); 56 ptr = NULL; 57 } 58 } get()59 webrtc::VoiceEngine* get() const { return ptr; } 60 private: 61 webrtc::VoiceEngine* ptr; 62 }; 63 64 // scoped_ptr class to handle obtaining and releasing WebRTC interface pointers 65 template<class T> 66 class scoped_voe_ptr { 67 public: scoped_voe_ptr(const scoped_voe_engine & e)68 explicit scoped_voe_ptr(const scoped_voe_engine& e) 69 : ptr(T::GetInterface(e.get())) {} scoped_voe_ptr(T * p)70 explicit scoped_voe_ptr(T* p) : ptr(p) {} ~scoped_voe_ptr()71 ~scoped_voe_ptr() { if (ptr) ptr->Release(); } 72 T* operator->() const { return ptr; } get()73 T* get() const { return ptr; } 74 75 // Releases the current pointer. reset()76 void reset() { 77 if (ptr) { 78 ptr->Release(); 79 ptr = NULL; 80 } 81 } 82 83 private: 84 T* ptr; 85 }; 86 87 // Utility class for aggregating the various WebRTC interface. 88 // Fake implementations can also be injected for testing. 89 class VoEWrapper { 90 public: VoEWrapper()91 VoEWrapper() 92 : engine_(webrtc::VoiceEngine::Create()), processing_(engine_), 93 base_(engine_), codec_(engine_), 94 hw_(engine_), network_(engine_), 95 rtp_(engine_), volume_(engine_) { 96 } VoEWrapper(webrtc::VoEAudioProcessing * processing,webrtc::VoEBase * base,webrtc::VoECodec * codec,webrtc::VoEHardware * hw,webrtc::VoENetwork * network,webrtc::VoERTP_RTCP * rtp,webrtc::VoEVolumeControl * volume)97 VoEWrapper(webrtc::VoEAudioProcessing* processing, 98 webrtc::VoEBase* base, 99 webrtc::VoECodec* codec, 100 webrtc::VoEHardware* hw, 101 webrtc::VoENetwork* network, 102 webrtc::VoERTP_RTCP* rtp, 103 webrtc::VoEVolumeControl* volume) 104 : engine_(NULL), 105 processing_(processing), 106 base_(base), 107 codec_(codec), 108 hw_(hw), 109 network_(network), 110 rtp_(rtp), 111 volume_(volume) { 112 } ~VoEWrapper()113 ~VoEWrapper() {} engine()114 webrtc::VoiceEngine* engine() const { return engine_.get(); } processing()115 webrtc::VoEAudioProcessing* processing() const { return processing_.get(); } base()116 webrtc::VoEBase* base() const { return base_.get(); } codec()117 webrtc::VoECodec* codec() const { return codec_.get(); } hw()118 webrtc::VoEHardware* hw() const { return hw_.get(); } network()119 webrtc::VoENetwork* network() const { return network_.get(); } rtp()120 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } volume()121 webrtc::VoEVolumeControl* volume() const { return volume_.get(); } error()122 int error() { return base_->LastError(); } 123 124 private: 125 scoped_voe_engine engine_; 126 scoped_voe_ptr<webrtc::VoEAudioProcessing> processing_; 127 scoped_voe_ptr<webrtc::VoEBase> base_; 128 scoped_voe_ptr<webrtc::VoECodec> codec_; 129 scoped_voe_ptr<webrtc::VoEHardware> hw_; 130 scoped_voe_ptr<webrtc::VoENetwork> network_; 131 scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_; 132 scoped_voe_ptr<webrtc::VoEVolumeControl> volume_; 133 }; 134 } // namespace cricket 135 136 #endif // TALK_MEDIA_WEBRTCVOE_H_ 137