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 29 #ifndef TALK_MEDIA_WEBRTCVOE_H_ 30 #define TALK_MEDIA_WEBRTCVOE_H_ 31 32 #include "talk/base/common.h" 33 #include "talk/media/webrtc/webrtccommon.h" 34 35 #include "webrtc/common_types.h" 36 #include "webrtc/modules/audio_device/include/audio_device.h" 37 #include "webrtc/voice_engine/include/voe_audio_processing.h" 38 #include "webrtc/voice_engine/include/voe_base.h" 39 #include "webrtc/voice_engine/include/voe_codec.h" 40 #include "webrtc/voice_engine/include/voe_dtmf.h" 41 #include "webrtc/voice_engine/include/voe_errors.h" 42 #include "webrtc/voice_engine/include/voe_external_media.h" 43 #include "webrtc/voice_engine/include/voe_file.h" 44 #include "webrtc/voice_engine/include/voe_hardware.h" 45 #include "webrtc/voice_engine/include/voe_neteq_stats.h" 46 #include "webrtc/voice_engine/include/voe_network.h" 47 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" 48 #include "webrtc/voice_engine/include/voe_video_sync.h" 49 #include "webrtc/voice_engine/include/voe_volume_control.h" 50 51 namespace cricket { 52 // automatically handles lifetime of WebRtc VoiceEngine 53 class scoped_voe_engine { 54 public: 55 explicit scoped_voe_engine(webrtc::VoiceEngine* e) : ptr(e) {} 56 // VERIFY, to ensure that there are no leaks at shutdown 57 ~scoped_voe_engine() { if (ptr) VERIFY(webrtc::VoiceEngine::Delete(ptr)); } 58 // Releases the current pointer. 59 void reset() { 60 if (ptr) { 61 VERIFY(webrtc::VoiceEngine::Delete(ptr)); 62 ptr = NULL; 63 } 64 } 65 webrtc::VoiceEngine* get() const { return ptr; } 66 private: 67 webrtc::VoiceEngine* ptr; 68 }; 69 70 // scoped_ptr class to handle obtaining and releasing WebRTC interface pointers 71 template<class T> 72 class scoped_voe_ptr { 73 public: 74 explicit scoped_voe_ptr(const scoped_voe_engine& e) 75 : ptr(T::GetInterface(e.get())) {} 76 explicit scoped_voe_ptr(T* p) : ptr(p) {} 77 ~scoped_voe_ptr() { if (ptr) ptr->Release(); } 78 T* operator->() const { return ptr; } 79 T* get() const { return ptr; } 80 81 // Releases the current pointer. 82 void reset() { 83 if (ptr) { 84 ptr->Release(); 85 ptr = NULL; 86 } 87 } 88 89 private: 90 T* ptr; 91 }; 92 93 // Utility class for aggregating the various WebRTC interface. 94 // Fake implementations can also be injected for testing. 95 class VoEWrapper { 96 public: 97 VoEWrapper() 98 : engine_(webrtc::VoiceEngine::Create()), processing_(engine_), 99 base_(engine_), codec_(engine_), dtmf_(engine_), file_(engine_), 100 hw_(engine_), media_(engine_), neteq_(engine_), network_(engine_), 101 rtp_(engine_), sync_(engine_), volume_(engine_) { 102 } 103 VoEWrapper(webrtc::VoEAudioProcessing* processing, 104 webrtc::VoEBase* base, 105 webrtc::VoECodec* codec, 106 webrtc::VoEDtmf* dtmf, 107 webrtc::VoEFile* file, 108 webrtc::VoEHardware* hw, 109 webrtc::VoEExternalMedia* media, 110 webrtc::VoENetEqStats* neteq, 111 webrtc::VoENetwork* network, 112 webrtc::VoERTP_RTCP* rtp, 113 webrtc::VoEVideoSync* sync, 114 webrtc::VoEVolumeControl* volume) 115 : engine_(NULL), 116 processing_(processing), 117 base_(base), 118 codec_(codec), 119 dtmf_(dtmf), 120 file_(file), 121 hw_(hw), 122 media_(media), 123 neteq_(neteq), 124 network_(network), 125 rtp_(rtp), 126 sync_(sync), 127 volume_(volume) { 128 } 129 ~VoEWrapper() {} 130 webrtc::VoiceEngine* engine() const { return engine_.get(); } 131 webrtc::VoEAudioProcessing* processing() const { return processing_.get(); } 132 webrtc::VoEBase* base() const { return base_.get(); } 133 webrtc::VoECodec* codec() const { return codec_.get(); } 134 webrtc::VoEDtmf* dtmf() const { return dtmf_.get(); } 135 webrtc::VoEFile* file() const { return file_.get(); } 136 webrtc::VoEHardware* hw() const { return hw_.get(); } 137 webrtc::VoEExternalMedia* media() const { return media_.get(); } 138 webrtc::VoENetEqStats* neteq() const { return neteq_.get(); } 139 webrtc::VoENetwork* network() const { return network_.get(); } 140 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } 141 webrtc::VoEVideoSync* sync() const { return sync_.get(); } 142 webrtc::VoEVolumeControl* volume() const { return volume_.get(); } 143 int error() { return base_->LastError(); } 144 145 private: 146 scoped_voe_engine engine_; 147 scoped_voe_ptr<webrtc::VoEAudioProcessing> processing_; 148 scoped_voe_ptr<webrtc::VoEBase> base_; 149 scoped_voe_ptr<webrtc::VoECodec> codec_; 150 scoped_voe_ptr<webrtc::VoEDtmf> dtmf_; 151 scoped_voe_ptr<webrtc::VoEFile> file_; 152 scoped_voe_ptr<webrtc::VoEHardware> hw_; 153 scoped_voe_ptr<webrtc::VoEExternalMedia> media_; 154 scoped_voe_ptr<webrtc::VoENetEqStats> neteq_; 155 scoped_voe_ptr<webrtc::VoENetwork> network_; 156 scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_; 157 scoped_voe_ptr<webrtc::VoEVideoSync> sync_; 158 scoped_voe_ptr<webrtc::VoEVolumeControl> volume_; 159 }; 160 161 // Adds indirection to static WebRtc functions, allowing them to be mocked. 162 class VoETraceWrapper { 163 public: 164 virtual ~VoETraceWrapper() {} 165 166 virtual int SetTraceFilter(const unsigned int filter) { 167 return webrtc::VoiceEngine::SetTraceFilter(filter); 168 } 169 virtual int SetTraceFile(const char* fileNameUTF8) { 170 return webrtc::VoiceEngine::SetTraceFile(fileNameUTF8); 171 } 172 virtual int SetTraceCallback(webrtc::TraceCallback* callback) { 173 return webrtc::VoiceEngine::SetTraceCallback(callback); 174 } 175 }; 176 177 } // namespace cricket 178 179 #endif // TALK_MEDIA_WEBRTCVOE_H_ 180
