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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
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_BASE_MEDIAENGINE_H_
29#define TALK_MEDIA_BASE_MEDIAENGINE_H_
30
31#ifdef OSX
32#include <CoreAudio/CoreAudio.h>
33#endif
34
henrike@webrtc.org28e20752013-07-10 00:45:36 +000035#include <string>
36#include <vector>
37
henrike@webrtc.org28e20752013-07-10 00:45:36 +000038#include "talk/media/base/codec.h"
39#include "talk/media/base/mediachannel.h"
40#include "talk/media/base/mediacommon.h"
41#include "talk/media/base/videocapturer.h"
42#include "talk/media/base/videocommon.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000043#include "talk/media/base/voiceprocessor.h"
44#include "talk/media/devices/devicemanager.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000045#include "webrtc/base/fileutils.h"
46#include "webrtc/base/sigslotrepeater.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000047
48#if defined(GOOGLE_CHROME_BUILD) || defined(CHROMIUM_BUILD)
49#define DISABLE_MEDIA_ENGINE_FACTORY
50#endif
51
Fredrik Solenberg709ed672015-09-15 12:26:33 +020052namespace webrtc {
53class Call;
54class VoiceEngine;
55}
56
henrike@webrtc.org28e20752013-07-10 00:45:36 +000057namespace cricket {
58
59class VideoCapturer;
60
61// MediaEngineInterface is an abstraction of a media engine which can be
62// subclassed to support different media componentry backends.
63// It supports voice and video operations in the same class to facilitate
64// proper synchronization between both media types.
65class MediaEngineInterface {
66 public:
henrike@webrtc.org28e20752013-07-10 00:45:36 +000067 virtual ~MediaEngineInterface() {}
68
69 // Initialization
70 // Starts the engine.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000071 virtual bool Init(rtc::Thread* worker_thread) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000072 // Shuts down the engine.
73 virtual void Terminate() = 0;
Fredrik Solenberg709ed672015-09-15 12:26:33 +020074 // TODO(solenberg): Remove once VoE API refactoring is done.
75 virtual webrtc::VoiceEngine* GetVoE() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000076
77 // MediaChannel creation
78 // Creates a voice media channel. Returns NULL on failure.
Fredrik Solenberg709ed672015-09-15 12:26:33 +020079 virtual VoiceMediaChannel* CreateChannel(
80 webrtc::Call* call,
81 const AudioOptions& options) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000082 // Creates a video media channel, paired with the specified voice channel.
83 // Returns NULL on failure.
buildbot@webrtc.org1ecbe452014-10-14 20:29:28 +000084 virtual VideoMediaChannel* CreateVideoChannel(
Fredrik Solenberg709ed672015-09-15 12:26:33 +020085 webrtc::Call* call,
86 const VideoOptions& options) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000087
henrike@webrtc.org28e20752013-07-10 00:45:36 +000088 // Configuration
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +000089 // Gets global audio options.
90 virtual AudioOptions GetAudioOptions() const = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000091 // Sets global audio options. "options" are from AudioOptions, above.
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +000092 virtual bool SetAudioOptions(const AudioOptions& options) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000093 // Sets the default (maximum) codec/resolution and encoder option to capture
94 // and encode video.
95 virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config)
96 = 0;
97
98 // Device selection
henrike@webrtc.org28e20752013-07-10 00:45:36 +000099 virtual bool SetSoundDevices(const Device* in_device,
100 const Device* out_device) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000101
102 // Device configuration
103 // Gets the current speaker volume, as a value between 0 and 255.
104 virtual bool GetOutputVolume(int* level) = 0;
105 // Sets the current speaker volume, as a value between 0 and 255.
106 virtual bool SetOutputVolume(int level) = 0;
107
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000108 // Gets the current microphone level, as a value between 0 and 10.
109 virtual int GetInputLevel() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000110
111 virtual const std::vector<AudioCodec>& audio_codecs() = 0;
112 virtual const std::vector<RtpHeaderExtension>&
113 audio_rtp_header_extensions() = 0;
114 virtual const std::vector<VideoCodec>& video_codecs() = 0;
115 virtual const std::vector<RtpHeaderExtension>&
116 video_rtp_header_extensions() = 0;
117
118 // Logging control
119 virtual void SetVoiceLogging(int min_sev, const char* filter) = 0;
120 virtual void SetVideoLogging(int min_sev, const char* filter) = 0;
121
wu@webrtc.orga9890802013-12-13 00:21:03 +0000122 // Starts AEC dump using existing file.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000123 virtual bool StartAecDump(rtc::PlatformFile file) = 0;
ivoc112a3d82015-10-16 02:22:18 -0700124
ivoc797ef122015-10-22 03:25:41 -0700125 // Stops recording AEC dump.
126 virtual void StopAecDump() = 0;
127
ivoc112a3d82015-10-16 02:22:18 -0700128 // Starts RtcEventLog using existing file.
129 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
130
131 // Stops recording an RtcEventLog.
132 virtual void StopRtcEventLog() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000133};
134
135
136#if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
137class MediaEngineFactory {
138 public:
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000139 typedef cricket::MediaEngineInterface* (*MediaEngineCreateFunction)();
140 // Creates a media engine, using either the compiled system default or the
141 // creation function specified in SetCreateFunction, if specified.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000142 static MediaEngineInterface* Create();
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000143 // Sets the function used when calling Create. If unset, the compiled system
144 // default will be used. Returns the old create function, or NULL if one
145 // wasn't set. Likewise, NULL can be used as the |function| parameter to
146 // reset to the default behavior.
147 static MediaEngineCreateFunction SetCreateFunction(
148 MediaEngineCreateFunction function);
149 private:
150 static MediaEngineCreateFunction create_function_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000151};
152#endif
153
154// CompositeMediaEngine constructs a MediaEngine from separate
155// voice and video engine classes.
156template<class VOICE, class VIDEO>
157class CompositeMediaEngine : public MediaEngineInterface {
158 public:
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000159 virtual ~CompositeMediaEngine() {}
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000160 virtual bool Init(rtc::Thread* worker_thread) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000161 if (!voice_.Init(worker_thread))
162 return false;
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200163 video_.Init();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000164 return true;
165 }
166 virtual void Terminate() {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000167 voice_.Terminate();
168 }
169
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200170 virtual webrtc::VoiceEngine* GetVoE() {
171 return voice_.GetVoE();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000172 }
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200173 virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
174 const AudioOptions& options) {
175 return voice_.CreateChannel(call, options);
176 }
177 virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call,
178 const VideoOptions& options) {
179 return video_.CreateChannel(call, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000180 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000181
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000182 virtual AudioOptions GetAudioOptions() const {
183 return voice_.GetOptions();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000184 }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000185 virtual bool SetAudioOptions(const AudioOptions& options) {
186 return voice_.SetOptions(options);
187 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000188 virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
189 return video_.SetDefaultEncoderConfig(config);
190 }
191
192 virtual bool SetSoundDevices(const Device* in_device,
193 const Device* out_device) {
194 return voice_.SetDevices(in_device, out_device);
195 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000196
197 virtual bool GetOutputVolume(int* level) {
198 return voice_.GetOutputVolume(level);
199 }
200 virtual bool SetOutputVolume(int level) {
201 return voice_.SetOutputVolume(level);
202 }
203
204 virtual int GetInputLevel() {
205 return voice_.GetInputLevel();
206 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000207 virtual const std::vector<AudioCodec>& audio_codecs() {
208 return voice_.codecs();
209 }
210 virtual const std::vector<RtpHeaderExtension>& audio_rtp_header_extensions() {
211 return voice_.rtp_header_extensions();
212 }
213 virtual const std::vector<VideoCodec>& video_codecs() {
214 return video_.codecs();
215 }
216 virtual const std::vector<RtpHeaderExtension>& video_rtp_header_extensions() {
217 return video_.rtp_header_extensions();
218 }
219
220 virtual void SetVoiceLogging(int min_sev, const char* filter) {
wu@webrtc.org967bfff2013-09-19 05:49:50 +0000221 voice_.SetLogging(min_sev, filter);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000222 }
223 virtual void SetVideoLogging(int min_sev, const char* filter) {
wu@webrtc.org967bfff2013-09-19 05:49:50 +0000224 video_.SetLogging(min_sev, filter);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000225 }
226
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000227 virtual bool StartAecDump(rtc::PlatformFile file) {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000228 return voice_.StartAecDump(file);
229 }
230
ivoc797ef122015-10-22 03:25:41 -0700231 virtual void StopAecDump() {
232 voice_.StopAecDump();
233 }
234
ivoc112a3d82015-10-16 02:22:18 -0700235 virtual bool StartRtcEventLog(rtc::PlatformFile file) {
236 return voice_.StartRtcEventLog(file);
237 }
238
239 virtual void StopRtcEventLog() { voice_.StopRtcEventLog(); }
240
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000241 protected:
242 VOICE voice_;
243 VIDEO video_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000244};
245
246// NullVoiceEngine can be used with CompositeMediaEngine in the case where only
247// a video engine is desired.
248class NullVoiceEngine {
249 public:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000250 bool Init(rtc::Thread* worker_thread) { return true; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000251 void Terminate() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000252 // If you need this to return an actual channel, use FakeMediaEngine instead.
Jelena Marusicc28a8962015-05-29 15:05:44 +0200253 VoiceMediaChannel* CreateChannel(const AudioOptions& options) {
254 return nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000255 }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000256 AudioOptions GetOptions() const { return AudioOptions(); }
257 bool SetOptions(const AudioOptions& options) { return true; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000258 bool SetDevices(const Device* in_device, const Device* out_device) {
259 return true;
260 }
261 bool GetOutputVolume(int* level) {
262 *level = 0;
263 return true;
264 }
265 bool SetOutputVolume(int level) { return true; }
266 int GetInputLevel() { return 0; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000267 const std::vector<AudioCodec>& codecs() { return codecs_; }
268 const std::vector<RtpHeaderExtension>& rtp_header_extensions() {
269 return rtp_header_extensions_;
270 }
271 void SetLogging(int min_sev, const char* filter) {}
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000272 bool StartAecDump(rtc::PlatformFile file) { return false; }
ivoc112a3d82015-10-16 02:22:18 -0700273 bool StartRtcEventLog(rtc::PlatformFile file) { return false; }
274 void StopRtcEventLog() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000275
276 private:
277 std::vector<AudioCodec> codecs_;
278 std::vector<RtpHeaderExtension> rtp_header_extensions_;
279};
280
281// NullVideoEngine can be used with CompositeMediaEngine in the case where only
282// a voice engine is desired.
283class NullVideoEngine {
284 public:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000285 bool Init(rtc::Thread* worker_thread) { return true; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000286 void Terminate() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000287 // If you need this to return an actual channel, use FakeMediaEngine instead.
288 VideoMediaChannel* CreateChannel(
Noah Richards99c2fe52015-04-10 14:32:39 -0700289 const VideoOptions& options,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000290 VoiceMediaChannel* voice_media_channel) {
291 return NULL;
292 }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000293 bool SetOptions(const VideoOptions& options) { return true; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000294 bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
295 return true;
296 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000297 const std::vector<VideoCodec>& codecs() { return codecs_; }
298 const std::vector<RtpHeaderExtension>& rtp_header_extensions() {
299 return rtp_header_extensions_;
300 }
301 void SetLogging(int min_sev, const char* filter) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000302
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000303 private:
304 std::vector<VideoCodec> codecs_;
305 std::vector<RtpHeaderExtension> rtp_header_extensions_;
306};
307
308typedef CompositeMediaEngine<NullVoiceEngine, NullVideoEngine> NullMediaEngine;
309
310enum DataChannelType {
311 DCT_NONE = 0,
312 DCT_RTP = 1,
313 DCT_SCTP = 2
314};
315
316class DataEngineInterface {
317 public:
318 virtual ~DataEngineInterface() {}
319 virtual DataMediaChannel* CreateChannel(DataChannelType type) = 0;
320 virtual const std::vector<DataCodec>& data_codecs() = 0;
321};
322
323} // namespace cricket
324
325#endif // TALK_MEDIA_BASE_MEDIAENGINE_H_