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source: webrtc/talk/media/webrtc/webrtcvoe.h @ 0:4bda6873e34c

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Last change on this file since 0:4bda6873e34c was 0:4bda6873e34c, checked in by Michael Prittie <mprittie@…>, 6 years ago

Scrubbed password for publication.

File size: 6.6 KB
Line 
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
51namespace cricket {
52// automatically handles lifetime of WebRtc VoiceEngine
53class 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
71template<class T>
72class 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.
95class 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.
162class 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_
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