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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: | |
49 explicit scoped_voe_engine(webrtc::VoiceEngine* e) : ptr(e) {} | |
50 // VERIFY, to ensure that there are no leaks at shutdown | |
51 ~scoped_voe_engine() { if (ptr) VERIFY(webrtc::VoiceEngine::Delete(ptr)); } | |
52 // Releases the current pointer. | |
53 void reset() { | |
54 if (ptr) { | |
55 VERIFY(webrtc::VoiceEngine::Delete(ptr)); | |
56 ptr = NULL; | |
57 } | |
58 } | |
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: | |
68 explicit scoped_voe_ptr(const scoped_voe_engine& e) | |
69 : ptr(T::GetInterface(e.get())) {} | |
70 explicit scoped_voe_ptr(T* p) : ptr(p) {} | |
71 ~scoped_voe_ptr() { if (ptr) ptr->Release(); } | |
72 T* operator->() const { return ptr; } | |
73 T* get() const { return ptr; } | |
74 | |
75 // Releases the current pointer. | |
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: | |
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 } | |
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 } | |
113 ~VoEWrapper() {} | |
114 webrtc::VoiceEngine* engine() const { return engine_.get(); } | |
115 webrtc::VoEAudioProcessing* processing() const { return processing_.get(); } | |
116 webrtc::VoEBase* base() const { return base_.get(); } | |
117 webrtc::VoECodec* codec() const { return codec_.get(); } | |
118 webrtc::VoEHardware* hw() const { return hw_.get(); } | |
119 webrtc::VoENetwork* network() const { return network_.get(); } | |
120 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } | |
121 webrtc::VoEVolumeControl* volume() const { return volume_.get(); } | |
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_ | |
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