2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
11 // This sub-API supports the following functionalities:
13 // - Enables full duplex VoIP sessions via RTP using G.711 (mu-Law or A-Law).
14 // - Initialization and termination.
15 // - Trace information on text files or via callbacks.
16 // - Multi-channel support (mixing, sending to multiple destinations etc.).
18 // To support other codecs than G.711, the VoECodec sub-API must be utilized.
20 // Usage example, omitting error checking:
22 // using namespace webrtc;
23 // VoiceEngine* voe = VoiceEngine::Create();
24 // VoEBase* base = VoEBase::GetInterface(voe);
26 // int ch = base->CreateChannel();
27 // base->StartPlayout(ch);
29 // base->DeleteChannel(ch);
32 // VoiceEngine::Delete(voe);
34 #ifndef WEBRTC_VOICE_ENGINE_VOE_BASE_H
35 #define WEBRTC_VOICE_ENGINE_VOE_BASE_H
37 #include "webrtc/common_types.h"
41 class AudioDeviceModule;
42 class AudioProcessing;
46 const int kVoEDefault = -1;
48 // VoiceEngineObserver
49 class WEBRTC_DLLEXPORT VoiceEngineObserver
52 // This method will be called after the occurrence of any runtime error
53 // code, or warning notification, when the observer interface has been
54 // installed using VoEBase::RegisterVoiceEngineObserver().
55 virtual void CallbackOnError(int channel, int errCode) = 0;
58 virtual ~VoiceEngineObserver() {}
62 class WEBRTC_DLLEXPORT VoiceEngine
65 // Creates a VoiceEngine object, which can then be used to acquire
66 // sub-APIs. Returns NULL on failure.
67 static VoiceEngine* Create();
68 static VoiceEngine* Create(const Config& config);
70 // Deletes a created VoiceEngine object and releases the utilized resources.
71 // Note that if there are outstanding references held via other interfaces,
72 // the voice engine instance will not actually be deleted until those
73 // references have been released.
74 static bool Delete(VoiceEngine*& voiceEngine);
76 // Specifies the amount and type of trace information which will be
77 // created by the VoiceEngine.
78 static int SetTraceFilter(unsigned int filter);
80 // Sets the name of the trace file and enables non-encrypted trace messages.
81 static int SetTraceFile(const char* fileNameUTF8,
82 bool addFileCounter = false);
84 // Installs the TraceCallback implementation to ensure that the user
85 // receives callbacks for generated trace messages.
86 static int SetTraceCallback(TraceCallback* callback);
88 #if !defined(WEBRTC_CHROMIUM_BUILD)
89 static int SetAndroidObjects(void* javaVM, void* env, void* context);
98 class WEBRTC_DLLEXPORT VoEBase
101 // Factory for the VoEBase sub-API. Increases an internal reference
102 // counter if successful. Returns NULL if the API is not supported or if
103 // construction fails.
104 static VoEBase* GetInterface(VoiceEngine* voiceEngine);
106 // Releases the VoEBase sub-API and decreases an internal reference
107 // counter. Returns the new reference count. This value should be zero
108 // for all sub-APIs before the VoiceEngine object can be safely deleted.
109 virtual int Release() = 0;
111 // Installs the observer class to enable runtime error control and
112 // warning notifications.
113 virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) = 0;
115 // Removes and disables the observer class for runtime error control
116 // and warning notifications.
117 virtual int DeRegisterVoiceEngineObserver() = 0;
119 // Initializes all common parts of the VoiceEngine; e.g. all
120 // encoders/decoders, the sound card and core receiving components.
121 // This method also makes it possible to install some user-defined external
123 // - The Audio Device Module (ADM) which implements all the audio layer
124 // functionality in a separate (reference counted) module.
125 // - The AudioProcessing module handles capture-side processing. VoiceEngine
126 // takes ownership of this object.
127 // If NULL is passed for any of these, VoiceEngine will create its own.
128 // TODO(ajm): Remove default NULLs.
129 virtual int Init(AudioDeviceModule* external_adm = NULL,
130 AudioProcessing* audioproc = NULL) = 0;
132 // Returns NULL before Init() is called.
133 virtual AudioProcessing* audio_processing() = 0;
135 // Terminates all VoiceEngine functions and releses allocated resources.
136 virtual int Terminate() = 0;
138 // Creates a new channel and allocates the required resources for it.
139 // One can use |config| to configure the channel. Currently that is used for
140 // choosing between ACM1 and ACM2, when creating Audio Coding Module.
141 virtual int CreateChannel() = 0;
142 virtual int CreateChannel(const Config& config) = 0;
144 // Deletes an existing channel and releases the utilized resources.
145 virtual int DeleteChannel(int channel) = 0;
147 // Prepares and initiates the VoiceEngine for reception of
148 // incoming RTP/RTCP packets on the specified |channel|.
149 virtual int StartReceive(int channel) = 0;
151 // Stops receiving incoming RTP/RTCP packets on the specified |channel|.
152 virtual int StopReceive(int channel) = 0;
154 // Starts forwarding the packets to the mixer/soundcard for a
155 // specified |channel|.
156 virtual int StartPlayout(int channel) = 0;
158 // Stops forwarding the packets to the mixer/soundcard for a
159 // specified |channel|.
160 virtual int StopPlayout(int channel) = 0;
162 // Starts sending packets to an already specified IP address and
163 // port number for a specified |channel|.
164 virtual int StartSend(int channel) = 0;
166 // Stops sending packets from a specified |channel|.
167 virtual int StopSend(int channel) = 0;
169 // Gets the version information for VoiceEngine and its components.
170 virtual int GetVersion(char version[1024]) = 0;
172 // Gets the last VoiceEngine error code.
173 virtual int LastError() = 0;
175 // TODO(xians): Make the interface pure virtual after libjingle
176 // implements the interface in its FakeWebRtcVoiceEngine.
177 virtual AudioTransport* audio_transport() { return NULL; }
179 // To be removed. Don't use.
180 virtual int SetOnHoldStatus(int channel, bool enable,
181 OnHoldModes mode = kHoldSendAndPlay) { return -1; }
182 virtual int GetOnHoldStatus(int channel, bool& enabled,
183 OnHoldModes& mode) { return -1; }
184 virtual int SetNetEQPlayoutMode(int channel, NetEqModes mode) { return -1; }
185 virtual int GetNetEQPlayoutMode(int channel,
186 NetEqModes& mode) { return -1; }
190 virtual ~VoEBase() {}
193 } // namespace webrtc
195 #endif // WEBRTC_VOICE_ENGINE_VOE_BASE_H