content/media/webrtc/MediaEngineWebRTC.cpp@44a2da4a2ab2
content/media/webrtc/MediaEngineWebRTC.cpp
Fri, 16 Jan 2015 04:50:19 +0100
- author
- Michael Schloh von Bennewitz <michael@schloh.com>
- date
- Fri, 16 Jan 2015 04:50:19 +0100
- branch
- TOR_BUG_9701
- changeset 13
- 44a2da4a2ab2
- permissions
- -rw-r--r--
Replace accessor implementation with direct member state manipulation, by
request https://trac.torproject.org/projects/tor/ticket/9701#comment:32
1 /* This Source Code Form is subject to the terms of the Mozilla Public
2 * License, v. 2.0. If a copy of the MPL was not distributed with this file,
3 * You can obtain one at http://mozilla.org/MPL/2.0/. */
5 #ifdef MOZ_LOGGING
6 #define FORCE_PR_LOG
7 #endif
9 #if defined(PR_LOG)
10 #error "This file must be #included before any IPDL-generated files or other files that #include prlog.h"
11 #endif
13 #include "nsIPrefService.h"
14 #include "nsIPrefBranch.h"
16 #include "CSFLog.h"
17 #include "prenv.h"
19 #ifdef PR_LOGGING
20 static PRLogModuleInfo*
21 GetUserMediaLog()
22 {
23 static PRLogModuleInfo *sLog;
24 if (!sLog)
25 sLog = PR_NewLogModule("GetUserMedia");
26 return sLog;
27 }
28 #endif
30 #include "MediaEngineWebRTC.h"
31 #include "ImageContainer.h"
32 #include "nsIComponentRegistrar.h"
33 #include "MediaEngineTabVideoSource.h"
34 #include "nsITabSource.h"
35 #include "MediaTrackConstraints.h"
37 #ifdef MOZ_WIDGET_ANDROID
38 #include "AndroidJNIWrapper.h"
39 #include "AndroidBridge.h"
40 #endif
42 #undef LOG
43 #define LOG(args) PR_LOG(GetUserMediaLog(), PR_LOG_DEBUG, args)
45 namespace mozilla {
47 MediaEngineWebRTC::MediaEngineWebRTC(MediaEnginePrefs &aPrefs)
48 : mMutex("mozilla::MediaEngineWebRTC")
49 , mVideoEngine(nullptr)
50 , mVoiceEngine(nullptr)
51 , mVideoEngineInit(false)
52 , mAudioEngineInit(false)
53 , mHasTabVideoSource(false)
54 {
55 #ifndef MOZ_B2G_CAMERA
56 nsCOMPtr<nsIComponentRegistrar> compMgr;
57 NS_GetComponentRegistrar(getter_AddRefs(compMgr));
58 if (compMgr) {
59 compMgr->IsContractIDRegistered(NS_TABSOURCESERVICE_CONTRACTID, &mHasTabVideoSource);
60 }
61 #else
62 AsyncLatencyLogger::Get()->AddRef();
63 #endif
64 // XXX
65 gFarendObserver = new AudioOutputObserver();
66 }
68 void
69 MediaEngineWebRTC::EnumerateVideoDevices(nsTArray<nsRefPtr<MediaEngineVideoSource> >* aVSources)
70 {
71 #ifdef MOZ_B2G_CAMERA
72 MutexAutoLock lock(mMutex);
74 /**
75 * We still enumerate every time, in case a new device was plugged in since
76 * the last call. TODO: Verify that WebRTC actually does deal with hotplugging
77 * new devices (with or without new engine creation) and accordingly adjust.
78 * Enumeration is not neccessary if GIPS reports the same set of devices
79 * for a given instance of the engine. Likewise, if a device was plugged out,
80 * mVideoSources must be updated.
81 */
82 int num = 0;
83 nsresult result;
84 result = ICameraControl::GetNumberOfCameras(num);
85 if (num <= 0 || result != NS_OK) {
86 return;
87 }
89 for (int i = 0; i < num; i++) {
90 nsCString cameraName;
91 result = ICameraControl::GetCameraName(i, cameraName);
92 if (result != NS_OK) {
93 continue;
94 }
96 nsRefPtr<MediaEngineWebRTCVideoSource> vSource;
97 NS_ConvertUTF8toUTF16 uuid(cameraName);
98 if (mVideoSources.Get(uuid, getter_AddRefs(vSource))) {
99 // We've already seen this device, just append.
100 aVSources->AppendElement(vSource.get());
101 } else {
102 vSource = new MediaEngineWebRTCVideoSource(i);
103 mVideoSources.Put(uuid, vSource); // Hashtable takes ownership.
104 aVSources->AppendElement(vSource);
105 }
106 }
108 return;
109 #else
110 ScopedCustomReleasePtr<webrtc::ViEBase> ptrViEBase;
111 ScopedCustomReleasePtr<webrtc::ViECapture> ptrViECapture;
113 // We spawn threads to handle gUM runnables, so we must protect the member vars
114 MutexAutoLock lock(mMutex);
116 #ifdef MOZ_WIDGET_ANDROID
117 jobject context = mozilla::AndroidBridge::Bridge()->GetGlobalContextRef();
119 // get the JVM
120 JavaVM *jvm = mozilla::AndroidBridge::Bridge()->GetVM();
122 if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
123 LOG(("VieCapture:SetAndroidObjects Failed"));
124 return;
125 }
126 #endif
127 if (!mVideoEngine) {
128 if (!(mVideoEngine = webrtc::VideoEngine::Create())) {
129 return;
130 }
131 }
133 PRLogModuleInfo *logs = GetWebRTCLogInfo();
134 if (!gWebrtcTraceLoggingOn && logs && logs->level > 0) {
135 // no need to a critical section or lock here
136 gWebrtcTraceLoggingOn = 1;
138 const char *file = PR_GetEnv("WEBRTC_TRACE_FILE");
139 if (!file) {
140 file = "WebRTC.log";
141 }
143 LOG(("%s Logging webrtc to %s level %d", __FUNCTION__, file, logs->level));
145 mVideoEngine->SetTraceFilter(logs->level);
146 mVideoEngine->SetTraceFile(file);
147 }
149 ptrViEBase = webrtc::ViEBase::GetInterface(mVideoEngine);
150 if (!ptrViEBase) {
151 return;
152 }
154 if (!mVideoEngineInit) {
155 if (ptrViEBase->Init() < 0) {
156 return;
157 }
158 mVideoEngineInit = true;
159 }
161 ptrViECapture = webrtc::ViECapture::GetInterface(mVideoEngine);
162 if (!ptrViECapture) {
163 return;
164 }
166 /**
167 * We still enumerate every time, in case a new device was plugged in since
168 * the last call. TODO: Verify that WebRTC actually does deal with hotplugging
169 * new devices (with or without new engine creation) and accordingly adjust.
170 * Enumeration is not neccessary if GIPS reports the same set of devices
171 * for a given instance of the engine. Likewise, if a device was plugged out,
172 * mVideoSources must be updated.
173 */
174 int num = ptrViECapture->NumberOfCaptureDevices();
175 if (num <= 0) {
176 return;
177 }
179 for (int i = 0; i < num; i++) {
180 const unsigned int kMaxDeviceNameLength = 128; // XXX FIX!
181 const unsigned int kMaxUniqueIdLength = 256;
182 char deviceName[kMaxDeviceNameLength];
183 char uniqueId[kMaxUniqueIdLength];
185 // paranoia
186 deviceName[0] = '\0';
187 uniqueId[0] = '\0';
188 int error = ptrViECapture->GetCaptureDevice(i, deviceName,
189 sizeof(deviceName), uniqueId,
190 sizeof(uniqueId));
192 if (error) {
193 LOG((" VieCapture:GetCaptureDevice: Failed %d",
194 ptrViEBase->LastError() ));
195 continue;
196 }
197 #ifdef DEBUG
198 LOG((" Capture Device Index %d, Name %s", i, deviceName));
200 webrtc::CaptureCapability cap;
201 int numCaps = ptrViECapture->NumberOfCapabilities(uniqueId, kMaxUniqueIdLength);
202 LOG(("Number of Capabilities %d", numCaps));
203 for (int j = 0; j < numCaps; j++) {
204 if (ptrViECapture->GetCaptureCapability(uniqueId, kMaxUniqueIdLength,
205 j, cap ) != 0 ) {
206 break;
207 }
208 LOG(("type=%d width=%d height=%d maxFPS=%d",
209 cap.rawType, cap.width, cap.height, cap.maxFPS ));
210 }
211 #endif
213 if (uniqueId[0] == '\0') {
214 // In case a device doesn't set uniqueId!
215 strncpy(uniqueId, deviceName, sizeof(uniqueId));
216 uniqueId[sizeof(uniqueId)-1] = '\0'; // strncpy isn't safe
217 }
219 nsRefPtr<MediaEngineWebRTCVideoSource> vSource;
220 NS_ConvertUTF8toUTF16 uuid(uniqueId);
221 if (mVideoSources.Get(uuid, getter_AddRefs(vSource))) {
222 // We've already seen this device, just append.
223 aVSources->AppendElement(vSource.get());
224 } else {
225 vSource = new MediaEngineWebRTCVideoSource(mVideoEngine, i);
226 mVideoSources.Put(uuid, vSource); // Hashtable takes ownership.
227 aVSources->AppendElement(vSource);
228 }
229 }
231 if (mHasTabVideoSource)
232 aVSources->AppendElement(new MediaEngineTabVideoSource());
234 return;
235 #endif
236 }
238 void
239 MediaEngineWebRTC::EnumerateAudioDevices(nsTArray<nsRefPtr<MediaEngineAudioSource> >* aASources)
240 {
241 ScopedCustomReleasePtr<webrtc::VoEBase> ptrVoEBase;
242 ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw;
243 // We spawn threads to handle gUM runnables, so we must protect the member vars
244 MutexAutoLock lock(mMutex);
246 #ifdef MOZ_WIDGET_ANDROID
247 jobject context = mozilla::AndroidBridge::Bridge()->GetGlobalContextRef();
249 // get the JVM
250 JavaVM *jvm = mozilla::AndroidBridge::Bridge()->GetVM();
251 JNIEnv *env = GetJNIForThread();
253 if (webrtc::VoiceEngine::SetAndroidObjects(jvm, env, (void*)context) != 0) {
254 LOG(("VoiceEngine:SetAndroidObjects Failed"));
255 return;
256 }
257 #endif
259 if (!mVoiceEngine) {
260 mVoiceEngine = webrtc::VoiceEngine::Create();
261 if (!mVoiceEngine) {
262 return;
263 }
264 }
266 PRLogModuleInfo *logs = GetWebRTCLogInfo();
267 if (!gWebrtcTraceLoggingOn && logs && logs->level > 0) {
268 // no need to a critical section or lock here
269 gWebrtcTraceLoggingOn = 1;
271 const char *file = PR_GetEnv("WEBRTC_TRACE_FILE");
272 if (!file) {
273 file = "WebRTC.log";
274 }
276 LOG(("Logging webrtc to %s level %d", __FUNCTION__, file, logs->level));
278 mVoiceEngine->SetTraceFilter(logs->level);
279 mVoiceEngine->SetTraceFile(file);
280 }
282 ptrVoEBase = webrtc::VoEBase::GetInterface(mVoiceEngine);
283 if (!ptrVoEBase) {
284 return;
285 }
287 if (!mAudioEngineInit) {
288 if (ptrVoEBase->Init() < 0) {
289 return;
290 }
291 mAudioEngineInit = true;
292 }
294 ptrVoEHw = webrtc::VoEHardware::GetInterface(mVoiceEngine);
295 if (!ptrVoEHw) {
296 return;
297 }
299 int nDevices = 0;
300 ptrVoEHw->GetNumOfRecordingDevices(nDevices);
301 for (int i = 0; i < nDevices; i++) {
302 // We use constants here because GetRecordingDeviceName takes char[128].
303 char deviceName[128];
304 char uniqueId[128];
305 // paranoia; jingle doesn't bother with this
306 deviceName[0] = '\0';
307 uniqueId[0] = '\0';
309 int error = ptrVoEHw->GetRecordingDeviceName(i, deviceName, uniqueId);
310 if (error) {
311 LOG((" VoEHardware:GetRecordingDeviceName: Failed %d",
312 ptrVoEBase->LastError() ));
313 continue;
314 }
316 if (uniqueId[0] == '\0') {
317 // Mac and Linux don't set uniqueId!
318 MOZ_ASSERT(sizeof(deviceName) == sizeof(uniqueId)); // total paranoia
319 strcpy(uniqueId,deviceName); // safe given assert and initialization/error-check
320 }
322 nsRefPtr<MediaEngineWebRTCAudioSource> aSource;
323 NS_ConvertUTF8toUTF16 uuid(uniqueId);
324 if (mAudioSources.Get(uuid, getter_AddRefs(aSource))) {
325 // We've already seen this device, just append.
326 aASources->AppendElement(aSource.get());
327 } else {
328 aSource = new MediaEngineWebRTCAudioSource(
329 mVoiceEngine, i, deviceName, uniqueId
330 );
331 mAudioSources.Put(uuid, aSource); // Hashtable takes ownership.
332 aASources->AppendElement(aSource);
333 }
334 }
335 }
337 void
338 MediaEngineWebRTC::Shutdown()
339 {
340 // This is likely paranoia
341 MutexAutoLock lock(mMutex);
343 if (mVideoEngine) {
344 mVideoSources.Clear();
345 webrtc::VideoEngine::Delete(mVideoEngine);
346 }
348 if (mVoiceEngine) {
349 mAudioSources.Clear();
350 webrtc::VoiceEngine::Delete(mVoiceEngine);
351 }
353 mVideoEngine = nullptr;
354 mVoiceEngine = nullptr;
355 }
357 }
