media/webrtc/signaling/src/media-conduit/VideoConduit.cpp@ac0c01689b40
media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
Thu, 15 Jan 2015 15:59:08 +0100
- author
- Michael Schloh von Bennewitz <michael@schloh.com>
- date
- Thu, 15 Jan 2015 15:59:08 +0100
- branch
- TOR_BUG_9701
- changeset 10
- ac0c01689b40
- permissions
- -rw-r--r--
Implement a real Private Browsing Mode condition by changing the API/ABI;
This solves Tor bug #9701, complying with disk avoidance documented in
https://www.torproject.org/projects/torbrowser/design/#disk-avoidance.
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 #include "CSFLog.h"
6 #include "nspr.h"
8 // For rtcp-fb constants
9 #include "ccsdp.h"
11 #include "VideoConduit.h"
12 #include "AudioConduit.h"
13 #include "nsThreadUtils.h"
14 #include "LoadManager.h"
15 #include "YuvStamper.h"
16 #include "nsServiceManagerUtils.h"
17 #include "nsIPrefService.h"
18 #include "nsIPrefBranch.h"
20 #include "webrtc/common_video/interface/native_handle.h"
21 #include "webrtc/video_engine/include/vie_errors.h"
23 #ifdef MOZ_WIDGET_ANDROID
24 #include "AndroidJNIWrapper.h"
25 #endif
27 #include <algorithm>
28 #include <math.h>
30 namespace mozilla {
32 static const char* logTag ="WebrtcVideoSessionConduit";
34 // 32 bytes is what WebRTC CodecInst expects
35 const unsigned int WebrtcVideoConduit::CODEC_PLNAME_SIZE = 32;
37 /**
38 * Factory Method for VideoConduit
39 */
40 mozilla::RefPtr<VideoSessionConduit> VideoSessionConduit::Create(VideoSessionConduit *aOther)
41 {
42 #ifdef MOZILLA_INTERNAL_API
43 // unit tests create their own "main thread"
44 NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
45 #endif
46 CSFLogDebug(logTag, "%s ", __FUNCTION__);
48 WebrtcVideoConduit* obj = new WebrtcVideoConduit();
49 if(obj->Init(static_cast<WebrtcVideoConduit*>(aOther)) != kMediaConduitNoError)
50 {
51 CSFLogError(logTag, "%s VideoConduit Init Failed ", __FUNCTION__);
52 delete obj;
53 return nullptr;
54 }
55 CSFLogDebug(logTag, "%s Successfully created VideoConduit ", __FUNCTION__);
56 return obj;
57 }
59 WebrtcVideoConduit::~WebrtcVideoConduit()
60 {
61 #ifdef MOZILLA_INTERNAL_API
62 // unit tests create their own "main thread"
63 NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
64 #endif
65 CSFLogDebug(logTag, "%s ", __FUNCTION__);
67 for(std::vector<VideoCodecConfig*>::size_type i=0;i < mRecvCodecList.size();i++)
68 {
69 delete mRecvCodecList[i];
70 }
72 delete mCurSendCodecConfig;
74 // The first one of a pair to be deleted shuts down media for both
75 //Deal with External Capturer
76 if(mPtrViECapture)
77 {
78 if (!mShutDown) {
79 mPtrViECapture->DisconnectCaptureDevice(mCapId);
80 mPtrViECapture->ReleaseCaptureDevice(mCapId);
81 mPtrExtCapture = nullptr;
82 if (mOtherDirection)
83 mOtherDirection->mPtrExtCapture = nullptr;
84 }
85 }
87 //Deal with External Renderer
88 if(mPtrViERender)
89 {
90 if (!mShutDown) {
91 if(mRenderer) {
92 mPtrViERender->StopRender(mChannel);
93 }
94 mPtrViERender->RemoveRenderer(mChannel);
95 }
96 }
98 //Deal with the transport
99 if(mPtrViENetwork)
100 {
101 if (!mShutDown) {
102 mPtrViENetwork->DeregisterSendTransport(mChannel);
103 }
104 }
106 if(mPtrViEBase)
107 {
108 if (!mShutDown) {
109 mPtrViEBase->StopSend(mChannel);
110 mPtrViEBase->StopReceive(mChannel);
111 SyncTo(nullptr);
112 mPtrViEBase->DeleteChannel(mChannel);
113 }
114 }
116 if (mOtherDirection)
117 {
118 // mOtherDirection owns these now!
119 mOtherDirection->mOtherDirection = nullptr;
120 // let other side we terminated the channel
121 mOtherDirection->mShutDown = true;
122 mVideoEngine = nullptr;
123 } else {
124 // We can't delete the VideoEngine until all these are released!
125 // And we can't use a Scoped ptr, since the order is arbitrary
126 mPtrViEBase = nullptr;
127 mPtrViECapture = nullptr;
128 mPtrViECodec = nullptr;
129 mPtrViENetwork = nullptr;
130 mPtrViERender = nullptr;
131 mPtrRTP = nullptr;
132 mPtrExtCodec = nullptr;
134 // only one opener can call Delete. Have it be the last to close.
135 if(mVideoEngine)
136 {
137 webrtc::VideoEngine::Delete(mVideoEngine);
138 }
139 }
140 }
142 bool WebrtcVideoConduit::GetLocalSSRC(unsigned int* ssrc) {
143 return !mPtrRTP->GetLocalSSRC(mChannel, *ssrc);
144 }
146 bool WebrtcVideoConduit::GetRemoteSSRC(unsigned int* ssrc) {
147 return !mPtrRTP->GetRemoteSSRC(mChannel, *ssrc);
148 }
150 bool WebrtcVideoConduit::GetAVStats(int32_t* jitterBufferDelayMs,
151 int32_t* playoutBufferDelayMs,
152 int32_t* avSyncOffsetMs) {
153 return false;
154 }
156 bool WebrtcVideoConduit::GetRTPStats(unsigned int* jitterMs,
157 unsigned int* cumulativeLost) {
158 unsigned short fractionLost;
159 unsigned extendedMax;
160 int rttMs;
161 // GetReceivedRTCPStatistics is a poorly named GetRTPStatistics variant
162 return !mPtrRTP->GetReceivedRTCPStatistics(mChannel, fractionLost,
163 *cumulativeLost,
164 extendedMax,
165 *jitterMs,
166 rttMs);
167 }
169 bool WebrtcVideoConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
170 uint32_t* jitterMs,
171 uint32_t* packetsReceived,
172 uint64_t* bytesReceived,
173 uint32_t* cumulativeLost,
174 int32_t* rttMs) {
175 uint32_t ntpHigh, ntpLow;
176 uint16_t fractionLost;
177 bool result = !mPtrRTP->GetRemoteRTCPReceiverInfo(mChannel, ntpHigh, ntpLow,
178 *packetsReceived,
179 *bytesReceived,
180 jitterMs,
181 &fractionLost,
182 cumulativeLost,
183 rttMs);
184 if (result) {
185 *timestamp = NTPtoDOMHighResTimeStamp(ntpHigh, ntpLow);
186 }
187 return result;
188 }
190 bool WebrtcVideoConduit::GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
191 unsigned int* packetsSent,
192 uint64_t* bytesSent) {
193 struct webrtc::SenderInfo senderInfo;
194 bool result = !mPtrRTP->GetRemoteRTCPSenderInfo(mChannel, &senderInfo);
195 if (result) {
196 *timestamp = NTPtoDOMHighResTimeStamp(senderInfo.NTP_timestamp_high,
197 senderInfo.NTP_timestamp_low);
198 *packetsSent = senderInfo.sender_packet_count;
199 *bytesSent = senderInfo.sender_octet_count;
200 }
201 return result;
202 }
204 /**
205 * Peforms intialization of the MANDATORY components of the Video Engine
206 */
207 MediaConduitErrorCode WebrtcVideoConduit::Init(WebrtcVideoConduit *other)
208 {
209 CSFLogDebug(logTag, "%s this=%p other=%p", __FUNCTION__, this, other);
211 if (other) {
212 MOZ_ASSERT(!other->mOtherDirection);
213 other->mOtherDirection = this;
214 mOtherDirection = other;
216 // only one can call ::Create()/GetVideoEngine()
217 MOZ_ASSERT(other->mVideoEngine);
218 mVideoEngine = other->mVideoEngine;
219 } else {
221 #ifdef MOZ_WIDGET_ANDROID
222 jobject context = jsjni_GetGlobalContextRef();
224 // get the JVM
225 JavaVM *jvm = jsjni_GetVM();
227 if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
228 CSFLogError(logTag, "%s: could not set Android objects", __FUNCTION__);
229 return kMediaConduitSessionNotInited;
230 }
231 #endif
233 // Per WebRTC APIs below function calls return nullptr on failure
234 if( !(mVideoEngine = webrtc::VideoEngine::Create()) )
235 {
236 CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);
237 return kMediaConduitSessionNotInited;
238 }
240 PRLogModuleInfo *logs = GetWebRTCLogInfo();
241 if (!gWebrtcTraceLoggingOn && logs && logs->level > 0) {
242 // no need to a critical section or lock here
243 gWebrtcTraceLoggingOn = 1;
245 const char *file = PR_GetEnv("WEBRTC_TRACE_FILE");
246 if (!file) {
247 file = "WebRTC.log";
248 }
249 CSFLogDebug(logTag, "%s Logging webrtc to %s level %d", __FUNCTION__,
250 file, logs->level);
251 mVideoEngine->SetTraceFilter(logs->level);
252 mVideoEngine->SetTraceFile(file);
253 }
254 }
256 if( !(mPtrViEBase = ViEBase::GetInterface(mVideoEngine)))
257 {
258 CSFLogError(logTag, "%s Unable to get video base interface ", __FUNCTION__);
259 return kMediaConduitSessionNotInited;
260 }
262 if( !(mPtrViECapture = ViECapture::GetInterface(mVideoEngine)))
263 {
264 CSFLogError(logTag, "%s Unable to get video capture interface", __FUNCTION__);
265 return kMediaConduitSessionNotInited;
266 }
268 if( !(mPtrViECodec = ViECodec::GetInterface(mVideoEngine)))
269 {
270 CSFLogError(logTag, "%s Unable to get video codec interface ", __FUNCTION__);
271 return kMediaConduitSessionNotInited;
272 }
274 if( !(mPtrViENetwork = ViENetwork::GetInterface(mVideoEngine)))
275 {
276 CSFLogError(logTag, "%s Unable to get video network interface ", __FUNCTION__);
277 return kMediaConduitSessionNotInited;
278 }
280 if( !(mPtrViERender = ViERender::GetInterface(mVideoEngine)))
281 {
282 CSFLogError(logTag, "%s Unable to get video render interface ", __FUNCTION__);
283 return kMediaConduitSessionNotInited;
284 }
286 if( !(mPtrRTP = webrtc::ViERTP_RTCP::GetInterface(mVideoEngine)))
287 {
288 CSFLogError(logTag, "%s Unable to get video RTCP interface ", __FUNCTION__);
289 return kMediaConduitSessionNotInited;
290 }
292 if ( !(mPtrExtCodec = webrtc::ViEExternalCodec::GetInterface(mVideoEngine)))
293 {
294 CSFLogError(logTag, "%s Unable to get external codec interface %d ",
295 __FUNCTION__, mPtrViEBase->LastError());
296 return kMediaConduitSessionNotInited;
297 }
299 if (other) {
300 mChannel = other->mChannel;
301 mPtrExtCapture = other->mPtrExtCapture;
302 mCapId = other->mCapId;
303 } else {
304 CSFLogDebug(logTag, "%s Engine Created: Init'ng the interfaces ",__FUNCTION__);
306 if(mPtrViEBase->Init() == -1)
307 {
308 CSFLogError(logTag, " %s Video Engine Init Failed %d ",__FUNCTION__,
309 mPtrViEBase->LastError());
310 return kMediaConduitSessionNotInited;
311 }
313 if(mPtrViEBase->CreateChannel(mChannel) == -1)
314 {
315 CSFLogError(logTag, " %s Channel creation Failed %d ",__FUNCTION__,
316 mPtrViEBase->LastError());
317 return kMediaConduitChannelError;
318 }
320 if(mPtrViENetwork->RegisterSendTransport(mChannel, *this) == -1)
321 {
322 CSFLogError(logTag, "%s ViENetwork Failed %d ", __FUNCTION__,
323 mPtrViEBase->LastError());
324 return kMediaConduitTransportRegistrationFail;
325 }
327 if(mPtrViECapture->AllocateExternalCaptureDevice(mCapId,
328 mPtrExtCapture) == -1)
329 {
330 CSFLogError(logTag, "%s Unable to Allocate capture module: %d ",
331 __FUNCTION__, mPtrViEBase->LastError());
332 return kMediaConduitCaptureError;
333 }
335 if(mPtrViECapture->ConnectCaptureDevice(mCapId,mChannel) == -1)
336 {
337 CSFLogError(logTag, "%s Unable to Connect capture module: %d ",
338 __FUNCTION__,mPtrViEBase->LastError());
339 return kMediaConduitCaptureError;
340 }
342 if(mPtrViERender->AddRenderer(mChannel,
343 webrtc::kVideoI420,
344 (webrtc::ExternalRenderer*) this) == -1)
345 {
346 CSFLogError(logTag, "%s Failed to added external renderer ", __FUNCTION__);
347 return kMediaConduitInvalidRenderer;
348 }
349 // Set up some parameters, per juberti. Set MTU.
350 if(mPtrViENetwork->SetMTU(mChannel, 1200) != 0)
351 {
352 CSFLogError(logTag, "%s MTU Failed %d ", __FUNCTION__,
353 mPtrViEBase->LastError());
354 return kMediaConduitMTUError;
355 }
356 // Turn on RTCP and loss feedback reporting.
357 if(mPtrRTP->SetRTCPStatus(mChannel, webrtc::kRtcpCompound_RFC4585) != 0)
358 {
359 CSFLogError(logTag, "%s RTCPStatus Failed %d ", __FUNCTION__,
360 mPtrViEBase->LastError());
361 return kMediaConduitRTCPStatusError;
362 }
363 }
365 CSFLogError(logTag, "%s Initialization Done", __FUNCTION__);
366 return kMediaConduitNoError;
367 }
369 void
370 WebrtcVideoConduit::SyncTo(WebrtcAudioConduit *aConduit)
371 {
372 CSFLogDebug(logTag, "%s Synced to %p", __FUNCTION__, aConduit);
374 // SyncTo(value) syncs to the AudioConduit, and if already synced replaces
375 // the current sync target. SyncTo(nullptr) cancels any existing sync and
376 // releases the strong ref to AudioConduit.
377 if (aConduit) {
378 mPtrViEBase->SetVoiceEngine(aConduit->GetVoiceEngine());
379 mPtrViEBase->ConnectAudioChannel(mChannel, aConduit->GetChannel());
380 // NOTE: this means the VideoConduit will keep the AudioConduit alive!
381 } else if ((mOtherDirection && mOtherDirection->mSyncedTo) || mSyncedTo) {
382 mPtrViEBase->DisconnectAudioChannel(mChannel);
383 mPtrViEBase->SetVoiceEngine(nullptr);
384 }
386 // Now manage the shared sync reference (ugly)
387 if (mSyncedTo || !mOtherDirection ) {
388 mSyncedTo = aConduit;
389 } else {
390 mOtherDirection->mSyncedTo = aConduit;
391 }
392 }
394 MediaConduitErrorCode
395 WebrtcVideoConduit::AttachRenderer(mozilla::RefPtr<VideoRenderer> aVideoRenderer)
396 {
397 CSFLogDebug(logTag, "%s ", __FUNCTION__);
399 //null renderer
400 if(!aVideoRenderer)
401 {
402 CSFLogError(logTag, "%s NULL Renderer", __FUNCTION__);
403 MOZ_ASSERT(PR_FALSE);
404 return kMediaConduitInvalidRenderer;
405 }
407 //Start Rendering if we haven't already
408 if(!mRenderer)
409 {
410 mRenderer = aVideoRenderer; // must be done before StartRender()
412 if(mPtrViERender->StartRender(mChannel) == -1)
413 {
414 CSFLogError(logTag, "%s Starting the Renderer Failed %d ", __FUNCTION__,
415 mPtrViEBase->LastError());
416 mRenderer = nullptr;
417 return kMediaConduitRendererFail;
418 }
419 } else {
420 //Assign the new renderer - overwrites if there is already one
421 mRenderer = aVideoRenderer;
422 }
424 return kMediaConduitNoError;
425 }
427 void
428 WebrtcVideoConduit::DetachRenderer()
429 {
430 if(mRenderer)
431 {
432 mPtrViERender->StopRender(mChannel);
433 mRenderer = nullptr;
434 }
435 }
437 MediaConduitErrorCode
438 WebrtcVideoConduit::AttachTransport(mozilla::RefPtr<TransportInterface> aTransport)
439 {
440 CSFLogDebug(logTag, "%s ", __FUNCTION__);
441 if(!aTransport)
442 {
443 CSFLogError(logTag, "%s NULL Transport", __FUNCTION__);
444 return kMediaConduitInvalidTransport;
445 }
446 // set the transport
447 mTransport = aTransport;
448 return kMediaConduitNoError;
449 }
451 /**
452 * Note: Setting the send-codec on the Video Engine will restart the encoder,
453 * sets up new SSRC and reset RTP_RTCP module with the new codec setting.
454 */
455 MediaConduitErrorCode
456 WebrtcVideoConduit::ConfigureSendMediaCodec(const VideoCodecConfig* codecConfig)
457 {
458 CSFLogDebug(logTag, "%s ", __FUNCTION__);
459 bool codecFound = false;
460 MediaConduitErrorCode condError = kMediaConduitNoError;
461 int error = 0; //webrtc engine errors
462 webrtc::VideoCodec video_codec;
463 std::string payloadName;
465 //validate basic params
466 if((condError = ValidateCodecConfig(codecConfig,true)) != kMediaConduitNoError)
467 {
468 return condError;
469 }
471 //Check if we have same codec already applied
472 if(CheckCodecsForMatch(mCurSendCodecConfig, codecConfig))
473 {
474 CSFLogDebug(logTag, "%s Codec has been applied already ", __FUNCTION__);
475 return kMediaConduitCodecInUse;
476 }
478 //transmitting already ?
479 if(mEngineTransmitting)
480 {
481 CSFLogDebug(logTag, "%s Engine Already Sending. Attemping to Stop ", __FUNCTION__);
482 if(mPtrViEBase->StopSend(mChannel) == -1)
483 {
484 CSFLogError(logTag, "%s StopSend() Failed %d ",__FUNCTION__,
485 mPtrViEBase->LastError());
486 return kMediaConduitUnknownError;
487 }
488 }
490 mEngineTransmitting = false;
492 if (codecConfig->mLoadManager) {
493 mPtrViEBase->RegisterCpuOveruseObserver(mChannel, codecConfig->mLoadManager);
494 mPtrViEBase->SetLoadManager(codecConfig->mLoadManager);
495 }
497 // we should be good here to set the new codec.
498 for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++)
499 {
500 if(0 == mPtrViECodec->GetCodec(idx, video_codec))
501 {
502 payloadName = video_codec.plName;
503 if(codecConfig->mName.compare(payloadName) == 0)
504 {
505 CodecConfigToWebRTCCodec(codecConfig, video_codec);
506 codecFound = true;
507 break;
508 }
509 }
510 }//for
512 if(codecFound == false)
513 {
514 CSFLogError(logTag, "%s Codec Mismatch ", __FUNCTION__);
515 return kMediaConduitInvalidSendCodec;
516 }
518 if(mPtrViECodec->SetSendCodec(mChannel, video_codec) == -1)
519 {
520 error = mPtrViEBase->LastError();
521 if(error == kViECodecInvalidCodec)
522 {
523 CSFLogError(logTag, "%s Invalid Send Codec", __FUNCTION__);
524 return kMediaConduitInvalidSendCodec;
525 }
526 CSFLogError(logTag, "%s SetSendCodec Failed %d ", __FUNCTION__,
527 mPtrViEBase->LastError());
528 return kMediaConduitUnknownError;
529 }
530 mSendingWidth = 0;
531 mSendingHeight = 0;
533 if(codecConfig->RtcpFbIsSet(SDP_RTCP_FB_NACK_BASIC)) {
534 CSFLogDebug(logTag, "Enabling NACK (send) for video stream\n");
535 if (mPtrRTP->SetNACKStatus(mChannel, true) != 0)
536 {
537 CSFLogError(logTag, "%s NACKStatus Failed %d ", __FUNCTION__,
538 mPtrViEBase->LastError());
539 return kMediaConduitNACKStatusError;
540 }
541 }
543 if(mPtrViEBase->StartSend(mChannel) == -1)
544 {
545 CSFLogError(logTag, "%s Start Send Error %d ", __FUNCTION__,
546 mPtrViEBase->LastError());
547 return kMediaConduitUnknownError;
548 }
550 //Copy the applied config for future reference.
551 delete mCurSendCodecConfig;
553 mCurSendCodecConfig = new VideoCodecConfig(*codecConfig);
555 mPtrRTP->SetRembStatus(mChannel, true, false);
557 // by now we should be successfully started the transmission
558 mEngineTransmitting = true;
559 return kMediaConduitNoError;
560 }
562 MediaConduitErrorCode
563 WebrtcVideoConduit::ConfigureRecvMediaCodecs(
564 const std::vector<VideoCodecConfig* >& codecConfigList)
565 {
566 CSFLogDebug(logTag, "%s ", __FUNCTION__);
567 MediaConduitErrorCode condError = kMediaConduitNoError;
568 int error = 0; //webrtc engine errors
569 bool success = false;
570 std::string payloadName;
572 // are we receiving already? If so, stop receiving and playout
573 // since we can't apply new recv codec when the engine is playing.
574 if(mEngineReceiving)
575 {
576 CSFLogDebug(logTag, "%s Engine Already Receiving . Attemping to Stop ", __FUNCTION__);
577 if(mPtrViEBase->StopReceive(mChannel) == -1)
578 {
579 error = mPtrViEBase->LastError();
580 if(error == kViEBaseUnknownError)
581 {
582 CSFLogDebug(logTag, "%s StopReceive() Success ", __FUNCTION__);
583 mEngineReceiving = false;
584 } else {
585 CSFLogError(logTag, "%s StopReceive() Failed %d ", __FUNCTION__,
586 mPtrViEBase->LastError());
587 return kMediaConduitUnknownError;
588 }
589 }
590 }
592 mEngineReceiving = false;
594 if(codecConfigList.empty())
595 {
596 CSFLogError(logTag, "%s Zero number of codecs to configure", __FUNCTION__);
597 return kMediaConduitMalformedArgument;
598 }
600 webrtc::ViEKeyFrameRequestMethod kf_request = webrtc::kViEKeyFrameRequestNone;
601 bool use_nack_basic = false;
603 //Try Applying the codecs in the list
604 // we treat as success if atleast one codec was applied and reception was
605 // started successfully.
606 for(std::vector<VideoCodecConfig*>::size_type i=0;i < codecConfigList.size();i++)
607 {
608 //if the codec param is invalid or diplicate, return error
609 if((condError = ValidateCodecConfig(codecConfigList[i],false)) != kMediaConduitNoError)
610 {
611 return condError;
612 }
614 // Check for the keyframe request type: PLI is preferred
615 // over FIR, and FIR is preferred over none.
616 if (codecConfigList[i]->RtcpFbIsSet(SDP_RTCP_FB_NACK_PLI))
617 {
618 kf_request = webrtc::kViEKeyFrameRequestPliRtcp;
619 } else if(kf_request == webrtc::kViEKeyFrameRequestNone &&
620 codecConfigList[i]->RtcpFbIsSet(SDP_RTCP_FB_CCM_FIR))
621 {
622 kf_request = webrtc::kViEKeyFrameRequestFirRtcp;
623 }
625 // Check whether NACK is requested
626 if(codecConfigList[i]->RtcpFbIsSet(SDP_RTCP_FB_NACK_BASIC))
627 {
628 use_nack_basic = true;
629 }
631 webrtc::VideoCodec video_codec;
633 mEngineReceiving = false;
634 memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
635 //Retrieve pre-populated codec structure for our codec.
636 for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++)
637 {
638 if(mPtrViECodec->GetCodec(idx, video_codec) == 0)
639 {
640 payloadName = video_codec.plName;
641 if(codecConfigList[i]->mName.compare(payloadName) == 0)
642 {
643 CodecConfigToWebRTCCodec(codecConfigList[i], video_codec);
644 if(mPtrViECodec->SetReceiveCodec(mChannel,video_codec) == -1)
645 {
646 CSFLogError(logTag, "%s Invalid Receive Codec %d ", __FUNCTION__,
647 mPtrViEBase->LastError());
648 } else {
649 CSFLogError(logTag, "%s Successfully Set the codec %s", __FUNCTION__,
650 codecConfigList[i]->mName.c_str());
651 if(CopyCodecToDB(codecConfigList[i]))
652 {
653 success = true;
654 } else {
655 CSFLogError(logTag,"%s Unable to updated Codec Database", __FUNCTION__);
656 return kMediaConduitUnknownError;
657 }
658 }
659 break; //we found a match
660 }
661 }
662 }//end for codeclist
664 }//end for
666 if(!success)
667 {
668 CSFLogError(logTag, "%s Setting Receive Codec Failed ", __FUNCTION__);
669 return kMediaConduitInvalidReceiveCodec;
670 }
672 // XXX Currently, we gather up all of the feedback types that the remote
673 // party indicated it supports for all video codecs and configure the entire
674 // conduit based on those capabilities. This is technically out of spec,
675 // as these values should be configured on a per-codec basis. However,
676 // the video engine only provides this API on a per-conduit basis, so that's
677 // how we have to do it. The approach of considering the remote capablities
678 // for the entire conduit to be a union of all remote codec capabilities
679 // (rather than the more conservative approach of using an intersection)
680 // is made to provide as many feedback mechanisms as are likely to be
681 // processed by the remote party (and should be relatively safe, since the
682 // remote party is required to ignore feedback types that it does not
683 // understand).
684 //
685 // Note that our configuration uses this union of remote capabilites as
686 // input to the configuration. It is not isomorphic to the configuration.
687 // For example, it only makes sense to have one frame request mechanism
688 // active at a time; so, if the remote party indicates more than one
689 // supported mechanism, we're only configuring the one we most prefer.
690 //
691 // See http://code.google.com/p/webrtc/issues/detail?id=2331
693 if (kf_request != webrtc::kViEKeyFrameRequestNone)
694 {
695 CSFLogDebug(logTag, "Enabling %s frame requests for video stream\n",
696 (kf_request == webrtc::kViEKeyFrameRequestPliRtcp ?
697 "PLI" : "FIR"));
698 if(mPtrRTP->SetKeyFrameRequestMethod(mChannel, kf_request) != 0)
699 {
700 CSFLogError(logTag, "%s KeyFrameRequest Failed %d ", __FUNCTION__,
701 mPtrViEBase->LastError());
702 return kMediaConduitKeyFrameRequestError;
703 }
704 }
706 switch (kf_request) {
707 case webrtc::kViEKeyFrameRequestNone:
708 mFrameRequestMethod = FrameRequestNone;
709 break;
710 case webrtc::kViEKeyFrameRequestPliRtcp:
711 mFrameRequestMethod = FrameRequestPli;
712 break;
713 case webrtc::kViEKeyFrameRequestFirRtcp:
714 mFrameRequestMethod = FrameRequestFir;
715 break;
716 default:
717 MOZ_ASSERT(PR_FALSE);
718 mFrameRequestMethod = FrameRequestUnknown;
719 }
721 if(use_nack_basic)
722 {
723 CSFLogDebug(logTag, "Enabling NACK (recv) for video stream\n");
724 if (mPtrRTP->SetNACKStatus(mChannel, true) != 0)
725 {
726 CSFLogError(logTag, "%s NACKStatus Failed %d ", __FUNCTION__,
727 mPtrViEBase->LastError());
728 return kMediaConduitNACKStatusError;
729 }
730 }
731 mUsingNackBasic = use_nack_basic;
733 //Start Receive on the video engine
734 if(mPtrViEBase->StartReceive(mChannel) == -1)
735 {
736 error = mPtrViEBase->LastError();
737 CSFLogError(logTag, "%s Start Receive Error %d ", __FUNCTION__, error);
740 return kMediaConduitUnknownError;
741 }
743 #ifdef MOZILLA_INTERNAL_API
744 if (NS_IsMainThread()) {
745 nsresult rv;
746 nsCOMPtr<nsIPrefService> prefs = do_GetService("@mozilla.org/preferences-service;1", &rv);
747 if (NS_SUCCEEDED(rv)) {
748 nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs);
750 if (branch) {
751 branch->GetBoolPref("media.video.test_latency", &mVideoLatencyTestEnable);
752 }
753 }
754 }
755 #endif
757 // by now we should be successfully started the reception
758 mPtrRTP->SetRembStatus(mChannel, false, true);
759 mEngineReceiving = true;
760 DumpCodecDB();
761 return kMediaConduitNoError;
762 }
764 // XXX we need to figure out how to feed back changes in preferred capture
765 // resolution to the getUserMedia source
766 bool
767 WebrtcVideoConduit::SelectSendResolution(unsigned short width,
768 unsigned short height)
769 {
770 // XXX This will do bandwidth-resolution adaptation as well - bug 877954
772 // Limit resolution to max-fs while keeping same aspect ratio as the
773 // incoming image.
774 if (mCurSendCodecConfig && mCurSendCodecConfig->mMaxFrameSize)
775 {
776 unsigned int cur_fs, max_width, max_height, mb_width, mb_height, mb_max;
778 mb_width = (width + 15) >> 4;
779 mb_height = (height + 15) >> 4;
781 cur_fs = mb_width * mb_height;
783 // Limit resolution to max_fs, but don't scale up.
784 if (cur_fs > mCurSendCodecConfig->mMaxFrameSize)
785 {
786 double scale_ratio;
788 scale_ratio = sqrt((double) mCurSendCodecConfig->mMaxFrameSize /
789 (double) cur_fs);
791 mb_width = mb_width * scale_ratio;
792 mb_height = mb_height * scale_ratio;
794 // Adjust mb_width and mb_height if they were truncated to zero.
795 if (mb_width == 0) {
796 mb_width = 1;
797 mb_height = std::min(mb_height, mCurSendCodecConfig->mMaxFrameSize);
798 }
799 if (mb_height == 0) {
800 mb_height = 1;
801 mb_width = std::min(mb_width, mCurSendCodecConfig->mMaxFrameSize);
802 }
803 }
805 // Limit width/height seperately to limit effect of extreme aspect ratios.
806 mb_max = (unsigned) sqrt(8 * (double) mCurSendCodecConfig->mMaxFrameSize);
808 max_width = 16 * std::min(mb_width, mb_max);
809 max_height = 16 * std::min(mb_height, mb_max);
811 if (width * max_height > max_width * height)
812 {
813 if (width > max_width)
814 {
815 // Due to the value is truncated to integer here and forced to even
816 // value later, adding 1 to improve accuracy.
817 height = max_width * height / width + 1;
818 width = max_width;
819 }
820 }
821 else
822 {
823 if (height > max_height)
824 {
825 // Due to the value is truncated to integer here and forced to even
826 // value later, adding 1 to improve accuracy.
827 width = max_height * width / height + 1;
828 height = max_height;
829 }
830 }
832 // Favor even multiples of pixels for width and height.
833 width = std::max(width & ~1, 2);
834 height = std::max(height & ~1, 2);
835 }
837 // Adapt to getUserMedia resolution changes
838 // check if we need to reconfigure the sending resolution
839 if (mSendingWidth != width || mSendingHeight != height)
840 {
841 // This will avoid us continually retrying this operation if it fails.
842 // If the resolution changes, we'll try again. In the meantime, we'll
843 // keep using the old size in the encoder.
844 mSendingWidth = width;
845 mSendingHeight = height;
847 // Get current vie codec.
848 webrtc::VideoCodec vie_codec;
849 int32_t err;
851 if ((err = mPtrViECodec->GetSendCodec(mChannel, vie_codec)) != 0)
852 {
853 CSFLogError(logTag, "%s: GetSendCodec failed, err %d", __FUNCTION__, err);
854 return false;
855 }
856 if (vie_codec.width != width || vie_codec.height != height)
857 {
858 vie_codec.width = width;
859 vie_codec.height = height;
861 if ((err = mPtrViECodec->SetSendCodec(mChannel, vie_codec)) != 0)
862 {
863 CSFLogError(logTag, "%s: SetSendCodec(%ux%u) failed, err %d",
864 __FUNCTION__, width, height, err);
865 return false;
866 }
867 CSFLogDebug(logTag, "%s: Encoder resolution changed to %ux%u",
868 __FUNCTION__, width, height);
869 } // else no change; mSendingWidth likely was 0
870 }
871 return true;
872 }
874 MediaConduitErrorCode
875 WebrtcVideoConduit::SetExternalSendCodec(int pltype,
876 VideoEncoder* encoder) {
877 int ret = mPtrExtCodec->RegisterExternalSendCodec(mChannel,
878 pltype,
879 static_cast<WebrtcVideoEncoder*>(encoder),
880 false);
881 return ret ? kMediaConduitInvalidSendCodec : kMediaConduitNoError;
882 }
884 MediaConduitErrorCode
885 WebrtcVideoConduit::SetExternalRecvCodec(int pltype,
886 VideoDecoder* decoder) {
887 int ret = mPtrExtCodec->RegisterExternalReceiveCodec(mChannel,
888 pltype,
889 static_cast<WebrtcVideoDecoder*>(decoder));
890 return ret ? kMediaConduitInvalidReceiveCodec : kMediaConduitNoError;
891 }
893 MediaConduitErrorCode
894 WebrtcVideoConduit::SendVideoFrame(unsigned char* video_frame,
895 unsigned int video_frame_length,
896 unsigned short width,
897 unsigned short height,
898 VideoType video_type,
899 uint64_t capture_time)
900 {
901 CSFLogDebug(logTag, "%s ", __FUNCTION__);
903 //check for the parameters sanity
904 if(!video_frame || video_frame_length == 0 ||
905 width == 0 || height == 0)
906 {
907 CSFLogError(logTag, "%s Invalid Parameters ",__FUNCTION__);
908 MOZ_ASSERT(PR_FALSE);
909 return kMediaConduitMalformedArgument;
910 }
912 webrtc::RawVideoType type;
913 switch (video_type) {
914 case kVideoI420:
915 type = webrtc::kVideoI420;
916 break;
917 case kVideoNV21:
918 type = webrtc::kVideoNV21;
919 break;
920 default:
921 CSFLogError(logTag, "%s VideoType Invalid. Only 1420 and NV21 Supported",__FUNCTION__);
922 MOZ_ASSERT(PR_FALSE);
923 return kMediaConduitMalformedArgument;
924 }
925 //Transmission should be enabled before we insert any frames.
926 if(!mEngineTransmitting)
927 {
928 CSFLogError(logTag, "%s Engine not transmitting ", __FUNCTION__);
929 return kMediaConduitSessionNotInited;
930 }
932 // enforce even width/height (paranoia)
933 MOZ_ASSERT(!(width & 1));
934 MOZ_ASSERT(!(height & 1));
936 if (!SelectSendResolution(width, height))
937 {
938 return kMediaConduitCaptureError;
939 }
941 //insert the frame to video engine in I420 format only
942 MOZ_ASSERT(mPtrExtCapture);
943 if(mPtrExtCapture->IncomingFrame(video_frame,
944 video_frame_length,
945 width, height,
946 type,
947 (unsigned long long)capture_time) == -1)
948 {
949 CSFLogError(logTag, "%s IncomingFrame Failed %d ", __FUNCTION__,
950 mPtrViEBase->LastError());
951 return kMediaConduitCaptureError;
952 }
954 CSFLogDebug(logTag, "%s Inserted a frame", __FUNCTION__);
955 return kMediaConduitNoError;
956 }
958 // Transport Layer Callbacks
959 MediaConduitErrorCode
960 WebrtcVideoConduit::ReceivedRTPPacket(const void *data, int len)
961 {
962 CSFLogDebug(logTag, "%s: Channel %d, Len %d ", __FUNCTION__, mChannel, len);
964 // Media Engine should be receiving already.
965 if(mEngineReceiving)
966 {
967 // let the engine know of a RTP packet to decode
968 if(mPtrViENetwork->ReceivedRTPPacket(mChannel,data,len) == -1)
969 {
970 int error = mPtrViEBase->LastError();
971 CSFLogError(logTag, "%s RTP Processing Failed %d ", __FUNCTION__, error);
972 if(error >= kViERtpRtcpInvalidChannelId && error <= kViERtpRtcpRtcpDisabled)
973 {
974 return kMediaConduitRTPProcessingFailed;
975 }
976 return kMediaConduitRTPRTCPModuleError;
977 }
978 } else {
979 CSFLogError(logTag, "Error: %s when not receiving", __FUNCTION__);
980 return kMediaConduitSessionNotInited;
981 }
983 return kMediaConduitNoError;
984 }
986 MediaConduitErrorCode
987 WebrtcVideoConduit::ReceivedRTCPPacket(const void *data, int len)
988 {
989 CSFLogDebug(logTag, " %s Channel %d, Len %d ", __FUNCTION__, mChannel, len);
991 //Media Engine should be receiving already
992 if(mEngineTransmitting)
993 {
994 if(mPtrViENetwork->ReceivedRTCPPacket(mChannel,data,len) == -1)
995 {
996 int error = mPtrViEBase->LastError();
997 CSFLogError(logTag, "%s RTP Processing Failed %d", __FUNCTION__, error);
998 if(error >= kViERtpRtcpInvalidChannelId && error <= kViERtpRtcpRtcpDisabled)
999 {
1000 return kMediaConduitRTPProcessingFailed;
1001 }
1002 return kMediaConduitRTPRTCPModuleError;
1003 }
1004 } else {
1005 CSFLogError(logTag, "Error: %s when not receiving", __FUNCTION__);
1006 return kMediaConduitSessionNotInited;
1007 }
1008 return kMediaConduitNoError;
1009 }
1011 //WebRTC::RTP Callback Implementation
1012 int WebrtcVideoConduit::SendPacket(int channel, const void* data, int len)
1013 {
1014 CSFLogDebug(logTag, "%s : channel %d len %d %s", __FUNCTION__, channel, len,
1015 (mEngineReceiving && mOtherDirection) ? "(using mOtherDirection)" : "");
1017 if (mEngineReceiving)
1018 {
1019 if (mOtherDirection)
1020 {
1021 return mOtherDirection->SendPacket(channel, data, len);
1022 }
1023 CSFLogDebug(logTag, "%s : Asked to send RTP without an RTP sender on channel %d",
1024 __FUNCTION__, channel);
1025 return -1;
1026 } else {
1027 if(mTransport && (mTransport->SendRtpPacket(data, len) == NS_OK))
1028 {
1029 CSFLogDebug(logTag, "%s Sent RTP Packet ", __FUNCTION__);
1030 return len;
1031 } else {
1032 CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);
1033 return -1;
1034 }
1035 }
1036 }
1038 int WebrtcVideoConduit::SendRTCPPacket(int channel, const void* data, int len)
1039 {
1040 CSFLogDebug(logTag, "%s : channel %d , len %d ", __FUNCTION__, channel,len);
1042 if (mEngineTransmitting)
1043 {
1044 if (mOtherDirection)
1045 {
1046 return mOtherDirection->SendRTCPPacket(channel, data, len);
1047 }
1048 }
1050 // We come here if we have only one pipeline/conduit setup,
1051 // such as for unidirectional streams.
1052 // We also end up here if we are receiving
1053 if(mTransport && mTransport->SendRtcpPacket(data, len) == NS_OK)
1054 {
1055 CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
1056 return len;
1057 } else {
1058 CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
1059 return -1;
1060 }
1061 }
1063 // WebRTC::ExternalMedia Implementation
1064 int
1065 WebrtcVideoConduit::FrameSizeChange(unsigned int width,
1066 unsigned int height,
1067 unsigned int numStreams)
1068 {
1069 CSFLogDebug(logTag, "%s ", __FUNCTION__);
1072 mReceivingWidth = width;
1073 mReceivingHeight = height;
1075 if(mRenderer)
1076 {
1077 mRenderer->FrameSizeChange(width, height, numStreams);
1078 return 0;
1079 }
1081 CSFLogError(logTag, "%s Renderer is NULL ", __FUNCTION__);
1082 return -1;
1083 }
1085 int
1086 WebrtcVideoConduit::DeliverFrame(unsigned char* buffer,
1087 int buffer_size,
1088 uint32_t time_stamp,
1089 int64_t render_time,
1090 void *handle)
1091 {
1092 CSFLogDebug(logTag, "%s Buffer Size %d", __FUNCTION__, buffer_size);
1094 if(mRenderer)
1095 {
1096 layers::Image* img = nullptr;
1097 // |handle| should be a webrtc::NativeHandle if available.
1098 if (handle) {
1099 webrtc::NativeHandle* native_h = static_cast<webrtc::NativeHandle*>(handle);
1100 // In the handle, there should be a layers::Image.
1101 img = static_cast<layers::Image*>(native_h->GetHandle());
1102 }
1104 if (mVideoLatencyTestEnable && mReceivingWidth && mReceivingHeight) {
1105 uint64_t now = PR_Now();
1106 uint64_t timestamp = 0;
1107 bool ok = YuvStamper::Decode(mReceivingWidth, mReceivingHeight, mReceivingWidth,
1108 buffer,
1109 reinterpret_cast<unsigned char*>(×tamp),
1110 sizeof(timestamp), 0, 0);
1111 if (ok) {
1112 VideoLatencyUpdate(now - timestamp);
1113 }
1114 }
1116 const ImageHandle img_h(img);
1117 mRenderer->RenderVideoFrame(buffer, buffer_size, time_stamp, render_time,
1118 img_h);
1119 return 0;
1120 }
1122 CSFLogError(logTag, "%s Renderer is NULL ", __FUNCTION__);
1123 return -1;
1124 }
1126 /**
1127 * Copy the codec passed into Conduit's database
1128 */
1130 void
1131 WebrtcVideoConduit::CodecConfigToWebRTCCodec(const VideoCodecConfig* codecInfo,
1132 webrtc::VideoCodec& cinst)
1133 {
1134 cinst.plType = codecInfo->mType;
1135 // leave width/height alone; they'll be overridden on the first frame
1136 if (codecInfo->mMaxFrameRate > 0)
1137 {
1138 cinst.maxFramerate = codecInfo->mMaxFrameRate;
1139 }
1140 cinst.minBitrate = 200;
1141 cinst.startBitrate = 300;
1142 cinst.maxBitrate = 2000;
1143 }
1145 //Copy the codec passed into Conduit's database
1146 bool
1147 WebrtcVideoConduit::CopyCodecToDB(const VideoCodecConfig* codecInfo)
1148 {
1149 VideoCodecConfig* cdcConfig = new VideoCodecConfig(*codecInfo);
1150 mRecvCodecList.push_back(cdcConfig);
1151 return true;
1152 }
1154 bool
1155 WebrtcVideoConduit::CheckCodecsForMatch(const VideoCodecConfig* curCodecConfig,
1156 const VideoCodecConfig* codecInfo) const
1157 {
1158 if(!curCodecConfig)
1159 {
1160 return false;
1161 }
1163 if(curCodecConfig->mType == codecInfo->mType &&
1164 curCodecConfig->mName.compare(codecInfo->mName) == 0 &&
1165 curCodecConfig->mMaxFrameSize == codecInfo->mMaxFrameSize &&
1166 curCodecConfig->mMaxFrameRate == codecInfo->mMaxFrameRate)
1167 {
1168 return true;
1169 }
1171 return false;
1172 }
1174 /**
1175 * Checks if the codec is already in Conduit's database
1176 */
1177 bool
1178 WebrtcVideoConduit::CheckCodecForMatch(const VideoCodecConfig* codecInfo) const
1179 {
1180 //the db should have atleast one codec
1181 for(std::vector<VideoCodecConfig*>::size_type i=0;i < mRecvCodecList.size();i++)
1182 {
1183 if(CheckCodecsForMatch(mRecvCodecList[i],codecInfo))
1184 {
1185 //match
1186 return true;
1187 }
1188 }
1189 //no match or empty local db
1190 return false;
1191 }
1193 /**
1194 * Perform validation on the codecConfig to be applied
1195 * Verifies if the codec is already applied.
1196 */
1197 MediaConduitErrorCode
1198 WebrtcVideoConduit::ValidateCodecConfig(const VideoCodecConfig* codecInfo,
1199 bool send) const
1200 {
1201 bool codecAppliedAlready = false;
1203 if(!codecInfo)
1204 {
1205 CSFLogError(logTag, "%s Null CodecConfig ", __FUNCTION__);
1206 return kMediaConduitMalformedArgument;
1207 }
1209 if((codecInfo->mName.empty()) ||
1210 (codecInfo->mName.length() >= CODEC_PLNAME_SIZE))
1211 {
1212 CSFLogError(logTag, "%s Invalid Payload Name Length ", __FUNCTION__);
1213 return kMediaConduitMalformedArgument;
1214 }
1216 //check if we have the same codec already applied
1217 if(send)
1218 {
1219 codecAppliedAlready = CheckCodecsForMatch(mCurSendCodecConfig,codecInfo);
1220 } else {
1221 codecAppliedAlready = CheckCodecForMatch(codecInfo);
1222 }
1224 if(codecAppliedAlready)
1225 {
1226 CSFLogDebug(logTag, "%s Codec %s Already Applied ", __FUNCTION__, codecInfo->mName.c_str());
1227 return kMediaConduitCodecInUse;
1228 }
1229 return kMediaConduitNoError;
1230 }
1232 void
1233 WebrtcVideoConduit::DumpCodecDB() const
1234 {
1235 for(std::vector<VideoCodecConfig*>::size_type i=0;i<mRecvCodecList.size();i++)
1236 {
1237 CSFLogDebug(logTag,"Payload Name: %s", mRecvCodecList[i]->mName.c_str());
1238 CSFLogDebug(logTag,"Payload Type: %d", mRecvCodecList[i]->mType);
1239 CSFLogDebug(logTag,"Payload Max Frame Size: %d", mRecvCodecList[i]->mMaxFrameSize);
1240 CSFLogDebug(logTag,"Payload Max Frame Rate: %d", mRecvCodecList[i]->mMaxFrameRate);
1241 }
1242 }
1244 void
1245 WebrtcVideoConduit::VideoLatencyUpdate(uint64_t newSample)
1246 {
1247 mVideoLatencyAvg = (sRoundingPadding * newSample + sAlphaNum * mVideoLatencyAvg) / sAlphaDen;
1248 }
1250 uint64_t
1251 WebrtcVideoConduit::MozVideoLatencyAvg()
1252 {
1253 return mVideoLatencyAvg / sRoundingPadding;
1254 }
1256 }// end namespace
