media/webrtc/signaling/src/media-conduit/AudioConduit.cpp@ac0c01689b40
media/webrtc/signaling/src/media-conduit/AudioConduit.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 #ifdef HAVE_NETINET_IN_H
9 #include <netinet/in.h>
10 #elif defined XP_WIN
11 #include <winsock2.h>
12 #endif
14 #include "AudioConduit.h"
15 #include "nsCOMPtr.h"
16 #include "mozilla/Services.h"
17 #include "nsServiceManagerUtils.h"
18 #include "nsIPrefService.h"
19 #include "nsIPrefBranch.h"
20 #include "nsThreadUtils.h"
21 #ifdef MOZILLA_INTERNAL_API
22 #include "Latency.h"
23 #include "mozilla/Telemetry.h"
24 #endif
26 #include "webrtc/voice_engine/include/voe_errors.h"
27 #include "webrtc/system_wrappers/interface/clock.h"
29 #ifdef MOZ_WIDGET_ANDROID
30 #include "AndroidJNIWrapper.h"
31 #endif
33 namespace mozilla {
35 static const char* logTag ="WebrtcAudioSessionConduit";
37 // 32 bytes is what WebRTC CodecInst expects
38 const unsigned int WebrtcAudioConduit::CODEC_PLNAME_SIZE = 32;
40 /**
41 * Factory Method for AudioConduit
42 */
43 mozilla::RefPtr<AudioSessionConduit> AudioSessionConduit::Create(AudioSessionConduit *aOther)
44 {
45 CSFLogDebug(logTag, "%s ", __FUNCTION__);
46 #ifdef MOZILLA_INTERNAL_API
47 // unit tests create their own "main thread"
48 NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
49 #endif
51 WebrtcAudioConduit* obj = new WebrtcAudioConduit();
52 if(obj->Init(static_cast<WebrtcAudioConduit*>(aOther)) != kMediaConduitNoError)
53 {
54 CSFLogError(logTag, "%s AudioConduit Init Failed ", __FUNCTION__);
55 delete obj;
56 return nullptr;
57 }
58 CSFLogDebug(logTag, "%s Successfully created AudioConduit ", __FUNCTION__);
59 return obj;
60 }
62 /**
63 * Destruction defines for our super-classes
64 */
65 WebrtcAudioConduit::~WebrtcAudioConduit()
66 {
67 #ifdef MOZILLA_INTERNAL_API
68 // unit tests create their own "main thread"
69 NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
70 #endif
72 CSFLogDebug(logTag, "%s ", __FUNCTION__);
73 for(std::vector<AudioCodecConfig*>::size_type i=0;i < mRecvCodecList.size();i++)
74 {
75 delete mRecvCodecList[i];
76 }
77 delete mCurSendCodecConfig;
79 // The first one of a pair to be deleted shuts down media for both
80 if(mPtrVoEXmedia)
81 {
82 if (!mShutDown) {
83 mPtrVoEXmedia->SetExternalRecordingStatus(false);
84 mPtrVoEXmedia->SetExternalPlayoutStatus(false);
85 }
86 }
88 //Deal with the transport
89 if(mPtrVoENetwork)
90 {
91 if (!mShutDown) {
92 mPtrVoENetwork->DeRegisterExternalTransport(mChannel);
93 }
94 }
96 if(mPtrVoEBase)
97 {
98 if (!mShutDown) {
99 mPtrVoEBase->StopPlayout(mChannel);
100 mPtrVoEBase->StopSend(mChannel);
101 mPtrVoEBase->StopReceive(mChannel);
102 mPtrVoEBase->DeleteChannel(mChannel);
103 mPtrVoEBase->Terminate();
104 }
105 }
107 if (mOtherDirection)
108 {
109 // mOtherDirection owns these now!
110 mOtherDirection->mOtherDirection = nullptr;
111 // let other side we terminated the channel
112 mOtherDirection->mShutDown = true;
113 mVoiceEngine = nullptr;
114 } else {
115 // We shouldn't delete the VoiceEngine until all these are released!
116 // And we can't use a Scoped ptr, since the order is arbitrary
117 mPtrVoENetwork = nullptr;
118 mPtrVoEBase = nullptr;
119 mPtrVoECodec = nullptr;
120 mPtrVoEXmedia = nullptr;
121 mPtrVoEProcessing = nullptr;
122 mPtrVoEVideoSync = nullptr;
123 mPtrVoERTP_RTCP = nullptr;
124 mPtrRTP = nullptr;
126 // only one opener can call Delete. Have it be the last to close.
127 if(mVoiceEngine)
128 {
129 webrtc::VoiceEngine::Delete(mVoiceEngine);
130 }
131 }
132 }
134 bool WebrtcAudioConduit::GetLocalSSRC(unsigned int* ssrc) {
135 return !mPtrRTP->GetLocalSSRC(mChannel, *ssrc);
136 }
138 bool WebrtcAudioConduit::GetRemoteSSRC(unsigned int* ssrc) {
139 return !mPtrRTP->GetRemoteSSRC(mChannel, *ssrc);
140 }
142 bool WebrtcAudioConduit::GetAVStats(int32_t* jitterBufferDelayMs,
143 int32_t* playoutBufferDelayMs,
144 int32_t* avSyncOffsetMs) {
145 return !mPtrVoEVideoSync->GetDelayEstimate(mChannel,
146 jitterBufferDelayMs,
147 playoutBufferDelayMs,
148 avSyncOffsetMs);
149 }
151 bool WebrtcAudioConduit::GetRTPStats(unsigned int* jitterMs,
152 unsigned int* cumulativeLost) {
153 unsigned int maxJitterMs = 0;
154 unsigned int discardedPackets;
155 *jitterMs = 0;
156 *cumulativeLost = 0;
157 return !mPtrRTP->GetRTPStatistics(mChannel, *jitterMs, maxJitterMs,
158 discardedPackets, *cumulativeLost);
159 }
161 DOMHighResTimeStamp
162 NTPtoDOMHighResTimeStamp(uint32_t ntpHigh, uint32_t ntpLow) {
163 return (uint32_t(ntpHigh - webrtc::kNtpJan1970) +
164 double(ntpLow) / webrtc::kMagicNtpFractionalUnit) * 1000;
165 }
167 bool WebrtcAudioConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
168 uint32_t* jitterMs,
169 uint32_t* packetsReceived,
170 uint64_t* bytesReceived,
171 uint32_t* cumulativeLost,
172 int32_t* rttMs) {
173 uint32_t ntpHigh, ntpLow;
174 uint16_t fractionLost;
175 bool result = !mPtrRTP->GetRemoteRTCPReceiverInfo(mChannel, ntpHigh, ntpLow,
176 *packetsReceived,
177 *bytesReceived,
178 *jitterMs,
179 fractionLost,
180 *cumulativeLost,
181 *rttMs);
182 if (result) {
183 *timestamp = NTPtoDOMHighResTimeStamp(ntpHigh, ntpLow);
184 }
185 return result;
186 }
188 bool WebrtcAudioConduit::GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
189 unsigned int* packetsSent,
190 uint64_t* bytesSent) {
191 struct webrtc::SenderInfo senderInfo;
192 bool result = !mPtrRTP->GetRemoteRTCPSenderInfo(mChannel, &senderInfo);
193 if (result) {
194 *timestamp = NTPtoDOMHighResTimeStamp(senderInfo.NTP_timestamp_high,
195 senderInfo.NTP_timestamp_low);
196 *packetsSent = senderInfo.sender_packet_count;
197 *bytesSent = senderInfo.sender_octet_count;
198 }
199 return result;
200 }
202 /*
203 * WebRTCAudioConduit Implementation
204 */
205 MediaConduitErrorCode WebrtcAudioConduit::Init(WebrtcAudioConduit *other)
206 {
207 CSFLogDebug(logTag, "%s this=%p other=%p", __FUNCTION__, this, other);
209 if (other) {
210 MOZ_ASSERT(!other->mOtherDirection);
211 other->mOtherDirection = this;
212 mOtherDirection = other;
214 // only one can call ::Create()/GetVoiceEngine()
215 MOZ_ASSERT(other->mVoiceEngine);
216 mVoiceEngine = other->mVoiceEngine;
217 } else {
218 #ifdef MOZ_WIDGET_ANDROID
219 jobject context = jsjni_GetGlobalContextRef();
221 // get the JVM
222 JavaVM *jvm = jsjni_GetVM();
223 JNIEnv* jenv = jsjni_GetJNIForThread();
225 if (webrtc::VoiceEngine::SetAndroidObjects(jvm, jenv, (void*)context) != 0) {
226 CSFLogError(logTag, "%s Unable to set Android objects", __FUNCTION__);
227 return kMediaConduitSessionNotInited;
228 }
229 #endif
231 // Per WebRTC APIs below function calls return nullptr on failure
232 if(!(mVoiceEngine = webrtc::VoiceEngine::Create()))
233 {
234 CSFLogError(logTag, "%s Unable to create voice engine", __FUNCTION__);
235 return kMediaConduitSessionNotInited;
236 }
238 PRLogModuleInfo *logs = GetWebRTCLogInfo();
239 if (!gWebrtcTraceLoggingOn && logs && logs->level > 0) {
240 // no need to a critical section or lock here
241 gWebrtcTraceLoggingOn = 1;
243 const char *file = PR_GetEnv("WEBRTC_TRACE_FILE");
244 if (!file) {
245 file = "WebRTC.log";
246 }
247 CSFLogDebug(logTag, "%s Logging webrtc to %s level %d", __FUNCTION__,
248 file, logs->level);
249 mVoiceEngine->SetTraceFilter(logs->level);
250 mVoiceEngine->SetTraceFile(file);
251 }
252 }
254 if(!(mPtrVoEBase = VoEBase::GetInterface(mVoiceEngine)))
255 {
256 CSFLogError(logTag, "%s Unable to initialize VoEBase", __FUNCTION__);
257 return kMediaConduitSessionNotInited;
258 }
260 if(!(mPtrVoENetwork = VoENetwork::GetInterface(mVoiceEngine)))
261 {
262 CSFLogError(logTag, "%s Unable to initialize VoENetwork", __FUNCTION__);
263 return kMediaConduitSessionNotInited;
264 }
266 if(!(mPtrVoECodec = VoECodec::GetInterface(mVoiceEngine)))
267 {
268 CSFLogError(logTag, "%s Unable to initialize VoEBCodec", __FUNCTION__);
269 return kMediaConduitSessionNotInited;
270 }
272 if(!(mPtrVoEProcessing = VoEAudioProcessing::GetInterface(mVoiceEngine)))
273 {
274 CSFLogError(logTag, "%s Unable to initialize VoEProcessing", __FUNCTION__);
275 return kMediaConduitSessionNotInited;
276 }
277 if(!(mPtrVoEXmedia = VoEExternalMedia::GetInterface(mVoiceEngine)))
278 {
279 CSFLogError(logTag, "%s Unable to initialize VoEExternalMedia", __FUNCTION__);
280 return kMediaConduitSessionNotInited;
281 }
282 if(!(mPtrVoERTP_RTCP = VoERTP_RTCP::GetInterface(mVoiceEngine)))
283 {
284 CSFLogError(logTag, "%s Unable to initialize VoERTP_RTCP", __FUNCTION__);
285 return kMediaConduitSessionNotInited;
286 }
288 if(!(mPtrVoEVideoSync = VoEVideoSync::GetInterface(mVoiceEngine)))
289 {
290 CSFLogError(logTag, "%s Unable to initialize VoEVideoSync", __FUNCTION__);
291 return kMediaConduitSessionNotInited;
292 }
293 if (!(mPtrRTP = webrtc::VoERTP_RTCP::GetInterface(mVoiceEngine)))
294 {
295 CSFLogError(logTag, "%s Unable to get audio RTP/RTCP interface ",
296 __FUNCTION__);
297 return kMediaConduitSessionNotInited;
298 }
300 if (other) {
301 mChannel = other->mChannel;
302 } else {
303 // init the engine with our audio device layer
304 if(mPtrVoEBase->Init() == -1)
305 {
306 CSFLogError(logTag, "%s VoiceEngine Base Not Initialized", __FUNCTION__);
307 return kMediaConduitSessionNotInited;
308 }
310 if( (mChannel = mPtrVoEBase->CreateChannel()) == -1)
311 {
312 CSFLogError(logTag, "%s VoiceEngine Channel creation failed",__FUNCTION__);
313 return kMediaConduitChannelError;
314 }
316 CSFLogDebug(logTag, "%s Channel Created %d ",__FUNCTION__, mChannel);
318 if(mPtrVoENetwork->RegisterExternalTransport(mChannel, *this) == -1)
319 {
320 CSFLogError(logTag, "%s VoiceEngine, External Transport Failed",__FUNCTION__);
321 return kMediaConduitTransportRegistrationFail;
322 }
324 if(mPtrVoEXmedia->SetExternalRecordingStatus(true) == -1)
325 {
326 CSFLogError(logTag, "%s SetExternalRecordingStatus Failed %d",__FUNCTION__,
327 mPtrVoEBase->LastError());
328 return kMediaConduitExternalPlayoutError;
329 }
331 if(mPtrVoEXmedia->SetExternalPlayoutStatus(true) == -1)
332 {
333 CSFLogError(logTag, "%s SetExternalPlayoutStatus Failed %d ",__FUNCTION__,
334 mPtrVoEBase->LastError());
335 return kMediaConduitExternalRecordingError;
336 }
337 CSFLogDebug(logTag , "%s AudioSessionConduit Initialization Done (%p)",__FUNCTION__, this);
338 }
339 return kMediaConduitNoError;
340 }
342 // AudioSessionConduit Implementation
343 MediaConduitErrorCode
344 WebrtcAudioConduit::AttachTransport(mozilla::RefPtr<TransportInterface> aTransport)
345 {
346 CSFLogDebug(logTag, "%s ", __FUNCTION__);
348 if(!aTransport)
349 {
350 CSFLogError(logTag, "%s NULL Transport", __FUNCTION__);
351 return kMediaConduitInvalidTransport;
352 }
353 // set the transport
354 mTransport = aTransport;
355 return kMediaConduitNoError;
356 }
358 MediaConduitErrorCode
359 WebrtcAudioConduit::ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig)
360 {
361 CSFLogDebug(logTag, "%s ", __FUNCTION__);
362 MediaConduitErrorCode condError = kMediaConduitNoError;
363 int error = 0;//webrtc engine errors
364 webrtc::CodecInst cinst;
366 //validate codec param
367 if((condError = ValidateCodecConfig(codecConfig, true)) != kMediaConduitNoError)
368 {
369 return condError;
370 }
372 //are we transmitting already, stop and apply the send codec
373 if(mEngineTransmitting)
374 {
375 CSFLogDebug(logTag, "%s Engine Already Sending. Attemping to Stop ", __FUNCTION__);
376 if(mPtrVoEBase->StopSend(mChannel) == -1)
377 {
378 CSFLogError(logTag, "%s StopSend() Failed %d ", __FUNCTION__,
379 mPtrVoEBase->LastError());
380 return kMediaConduitUnknownError;
381 }
382 }
384 mEngineTransmitting = false;
386 if(!CodecConfigToWebRTCCodec(codecConfig,cinst))
387 {
388 CSFLogError(logTag,"%s CodecConfig to WebRTC Codec Failed ",__FUNCTION__);
389 return kMediaConduitMalformedArgument;
390 }
392 if(mPtrVoECodec->SetSendCodec(mChannel, cinst) == -1)
393 {
394 error = mPtrVoEBase->LastError();
395 CSFLogError(logTag, "%s SetSendCodec - Invalid Codec %d ",__FUNCTION__,
396 error);
398 if(error == VE_CANNOT_SET_SEND_CODEC || error == VE_CODEC_ERROR)
399 {
400 CSFLogError(logTag, "%s Invalid Send Codec", __FUNCTION__);
401 return kMediaConduitInvalidSendCodec;
402 }
403 CSFLogError(logTag, "%s SetSendCodec Failed %d ", __FUNCTION__,
404 mPtrVoEBase->LastError());
405 return kMediaConduitUnknownError;
406 }
408 #ifdef MOZILLA_INTERNAL_API
409 // TEMPORARY - see bug 694814 comment 2
410 nsresult rv;
411 nsCOMPtr<nsIPrefService> prefs = do_GetService("@mozilla.org/preferences-service;1", &rv);
412 if (NS_SUCCEEDED(rv)) {
413 nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs);
415 if (branch) {
416 branch->GetIntPref("media.peerconnection.capture_delay", &mCaptureDelay);
417 }
418 }
419 #endif
421 //Let's Send Transport State-machine on the Engine
422 if(mPtrVoEBase->StartSend(mChannel) == -1)
423 {
424 error = mPtrVoEBase->LastError();
425 CSFLogError(logTag, "%s StartSend failed %d", __FUNCTION__, error);
426 return kMediaConduitUnknownError;
427 }
429 //Copy the applied config for future reference.
430 delete mCurSendCodecConfig;
432 mCurSendCodecConfig = new AudioCodecConfig(codecConfig->mType,
433 codecConfig->mName,
434 codecConfig->mFreq,
435 codecConfig->mPacSize,
436 codecConfig->mChannels,
437 codecConfig->mRate,
438 codecConfig->mLoadManager);
440 mEngineTransmitting = true;
441 return kMediaConduitNoError;
442 }
444 MediaConduitErrorCode
445 WebrtcAudioConduit::ConfigureRecvMediaCodecs(
446 const std::vector<AudioCodecConfig*>& codecConfigList)
447 {
448 CSFLogDebug(logTag, "%s ", __FUNCTION__);
449 MediaConduitErrorCode condError = kMediaConduitNoError;
450 int error = 0; //webrtc engine errors
451 bool success = false;
453 // Are we receiving already? If so, stop receiving and playout
454 // since we can't apply new recv codec when the engine is playing.
455 if(mEngineReceiving)
456 {
457 CSFLogDebug(logTag, "%s Engine Already Receiving. Attemping to Stop ", __FUNCTION__);
458 // AudioEngine doesn't fail fatally on stopping reception. Ref:voe_errors.h.
459 // hence we need not be strict in failing here on errors
460 mPtrVoEBase->StopReceive(mChannel);
461 CSFLogDebug(logTag, "%s Attemping to Stop playout ", __FUNCTION__);
462 if(mPtrVoEBase->StopPlayout(mChannel) == -1)
463 {
464 if( mPtrVoEBase->LastError() == VE_CANNOT_STOP_PLAYOUT)
465 {
466 CSFLogDebug(logTag, "%s Stop-Playout Failed %d", __FUNCTION__, mPtrVoEBase->LastError());
467 return kMediaConduitPlayoutError;
468 }
469 }
470 }
472 mEngineReceiving = false;
474 if(codecConfigList.empty())
475 {
476 CSFLogError(logTag, "%s Zero number of codecs to configure", __FUNCTION__);
477 return kMediaConduitMalformedArgument;
478 }
480 // Try Applying the codecs in the list.
481 // We succeed if at least one codec was applied and reception was
482 // started successfully.
483 for(std::vector<AudioCodecConfig*>::size_type i=0 ;i<codecConfigList.size();i++)
484 {
485 //if the codec param is invalid or diplicate, return error
486 if((condError = ValidateCodecConfig(codecConfigList[i],false)) != kMediaConduitNoError)
487 {
488 return condError;
489 }
491 webrtc::CodecInst cinst;
492 if(!CodecConfigToWebRTCCodec(codecConfigList[i],cinst))
493 {
494 CSFLogError(logTag,"%s CodecConfig to WebRTC Codec Failed ",__FUNCTION__);
495 continue;
496 }
498 if(mPtrVoECodec->SetRecPayloadType(mChannel,cinst) == -1)
499 {
500 error = mPtrVoEBase->LastError();
501 CSFLogError(logTag, "%s SetRecvCodec Failed %d ",__FUNCTION__, error);
502 continue;
503 } else {
504 CSFLogDebug(logTag, "%s Successfully Set RecvCodec %s", __FUNCTION__,
505 codecConfigList[i]->mName.c_str());
506 //copy this to local database
507 if(CopyCodecToDB(codecConfigList[i]))
508 {
509 success = true;
510 } else {
511 CSFLogError(logTag,"%s Unable to updated Codec Database", __FUNCTION__);
512 return kMediaConduitUnknownError;
513 }
515 }
517 } //end for
519 if(!success)
520 {
521 CSFLogError(logTag, "%s Setting Receive Codec Failed ", __FUNCTION__);
522 return kMediaConduitInvalidReceiveCodec;
523 }
525 //If we are here, atleast one codec should have been set
526 if(mPtrVoEBase->StartReceive(mChannel) == -1)
527 {
528 error = mPtrVoEBase->LastError();
529 CSFLogError(logTag , "%s StartReceive Failed %d ",__FUNCTION__, error);
530 if(error == VE_RECV_SOCKET_ERROR)
531 {
532 return kMediaConduitSocketError;
533 }
534 return kMediaConduitUnknownError;
535 }
538 if(mPtrVoEBase->StartPlayout(mChannel) == -1)
539 {
540 CSFLogError(logTag, "%s Starting playout Failed", __FUNCTION__);
541 return kMediaConduitPlayoutError;
542 }
543 //we should be good here for setting this.
544 mEngineReceiving = true;
545 DumpCodecDB();
546 return kMediaConduitNoError;
547 }
548 MediaConduitErrorCode
549 WebrtcAudioConduit::EnableAudioLevelExtension(bool enabled, uint8_t id)
550 {
551 CSFLogDebug(logTag, "%s %d %d ", __FUNCTION__, enabled, id);
553 if (mPtrVoERTP_RTCP->SetRTPAudioLevelIndicationStatus(mChannel, enabled, id) == -1)
554 {
555 CSFLogError(logTag, "%s SetRTPAudioLevelIndicationStatus Failed", __FUNCTION__);
556 return kMediaConduitUnknownError;
557 }
559 return kMediaConduitNoError;
560 }
562 MediaConduitErrorCode
563 WebrtcAudioConduit::SendAudioFrame(const int16_t audio_data[],
564 int32_t lengthSamples,
565 int32_t samplingFreqHz,
566 int32_t capture_delay)
567 {
568 CSFLogDebug(logTag, "%s ", __FUNCTION__);
569 // Following checks need to be performed
570 // 1. Non null audio buffer pointer,
571 // 2. invalid sampling frequency - less than 0 or unsupported ones
572 // 3. Appropriate Sample Length for 10 ms audio-frame. This represents
573 // block size the VoiceEngine feeds into encoder for passed in audio-frame
574 // Ex: for 16000 sampling rate , valid block-length is 160
575 // Similarly for 32000 sampling rate, valid block length is 320
576 // We do the check by the verify modular operator below to be zero
578 if(!audio_data || (lengthSamples <= 0) ||
579 (IsSamplingFreqSupported(samplingFreqHz) == false) ||
580 ((lengthSamples % (samplingFreqHz / 100) != 0)) )
581 {
582 CSFLogError(logTag, "%s Invalid Parameters ",__FUNCTION__);
583 MOZ_ASSERT(PR_FALSE);
584 return kMediaConduitMalformedArgument;
585 }
587 //validate capture time
588 if(capture_delay < 0 )
589 {
590 CSFLogError(logTag,"%s Invalid Capture Delay ", __FUNCTION__);
591 MOZ_ASSERT(PR_FALSE);
592 return kMediaConduitMalformedArgument;
593 }
595 // if transmission is not started .. conduit cannot insert frames
596 if(!mEngineTransmitting)
597 {
598 CSFLogError(logTag, "%s Engine not transmitting ", __FUNCTION__);
599 return kMediaConduitSessionNotInited;
600 }
602 #ifdef MOZILLA_INTERNAL_API
603 if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
604 struct Processing insert = { TimeStamp::Now(), 0 };
605 mProcessing.AppendElement(insert);
606 }
607 #endif
609 capture_delay = mCaptureDelay;
610 //Insert the samples
611 if(mPtrVoEXmedia->ExternalRecordingInsertData(audio_data,
612 lengthSamples,
613 samplingFreqHz,
614 capture_delay) == -1)
615 {
616 int error = mPtrVoEBase->LastError();
617 CSFLogError(logTag, "%s Inserting audio data Failed %d", __FUNCTION__, error);
618 if(error == VE_RUNTIME_REC_ERROR)
619 {
620 return kMediaConduitRecordingError;
621 }
622 return kMediaConduitUnknownError;
623 }
624 // we should be good here
625 return kMediaConduitNoError;
626 }
628 MediaConduitErrorCode
629 WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
630 int32_t samplingFreqHz,
631 int32_t capture_delay,
632 int& lengthSamples)
633 {
635 CSFLogDebug(logTag, "%s ", __FUNCTION__);
636 unsigned int numSamples = 0;
638 //validate params
639 if(!speechData )
640 {
641 CSFLogError(logTag,"%s Null Audio Buffer Pointer", __FUNCTION__);
642 MOZ_ASSERT(PR_FALSE);
643 return kMediaConduitMalformedArgument;
644 }
646 // Validate sample length
647 if((numSamples = GetNum10msSamplesForFrequency(samplingFreqHz)) == 0 )
648 {
649 CSFLogError(logTag,"%s Invalid Sampling Frequency ", __FUNCTION__);
650 MOZ_ASSERT(PR_FALSE);
651 return kMediaConduitMalformedArgument;
652 }
654 //validate capture time
655 if(capture_delay < 0 )
656 {
657 CSFLogError(logTag,"%s Invalid Capture Delay ", __FUNCTION__);
658 MOZ_ASSERT(PR_FALSE);
659 return kMediaConduitMalformedArgument;
660 }
662 //Conduit should have reception enabled before we ask for decoded
663 // samples
664 if(!mEngineReceiving)
665 {
666 CSFLogError(logTag, "%s Engine not Receiving ", __FUNCTION__);
667 return kMediaConduitSessionNotInited;
668 }
671 lengthSamples = 0; //output paramter
673 if(mPtrVoEXmedia->ExternalPlayoutGetData( speechData,
674 samplingFreqHz,
675 capture_delay,
676 lengthSamples) == -1)
677 {
678 int error = mPtrVoEBase->LastError();
679 CSFLogError(logTag, "%s Getting audio data Failed %d", __FUNCTION__, error);
680 if(error == VE_RUNTIME_PLAY_ERROR)
681 {
682 return kMediaConduitPlayoutError;
683 }
684 return kMediaConduitUnknownError;
685 }
687 // Not #ifdef DEBUG or on a log module so we can use it for about:webrtc/etc
688 mSamples += lengthSamples;
689 if (mSamples >= mLastSyncLog + samplingFreqHz) {
690 int jitter_buffer_delay_ms;
691 int playout_buffer_delay_ms;
692 int avsync_offset_ms;
693 if (GetAVStats(&jitter_buffer_delay_ms,
694 &playout_buffer_delay_ms,
695 &avsync_offset_ms)) {
696 #ifdef MOZILLA_INTERNAL_API
697 if (avsync_offset_ms < 0) {
698 Telemetry::Accumulate(Telemetry::WEBRTC_AVSYNC_WHEN_VIDEO_LAGS_AUDIO_MS,
699 -avsync_offset_ms);
700 } else {
701 Telemetry::Accumulate(Telemetry::WEBRTC_AVSYNC_WHEN_AUDIO_LAGS_VIDEO_MS,
702 avsync_offset_ms);
703 }
704 #endif
705 CSFLogError(logTag,
706 "A/V sync: sync delta: %dms, audio jitter delay %dms, playout delay %dms",
707 avsync_offset_ms, jitter_buffer_delay_ms, playout_buffer_delay_ms);
708 } else {
709 CSFLogError(logTag, "A/V sync: GetAVStats failed");
710 }
711 mLastSyncLog = mSamples;
712 }
714 #ifdef MOZILLA_INTERNAL_API
715 if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
716 if (mProcessing.Length() > 0) {
717 unsigned int now;
718 mPtrVoEVideoSync->GetPlayoutTimestamp(mChannel, now);
719 if (static_cast<uint32_t>(now) != mLastTimestamp) {
720 mLastTimestamp = static_cast<uint32_t>(now);
721 // Find the block that includes this timestamp in the network input
722 while (mProcessing.Length() > 0) {
723 // FIX! assumes 20ms @ 48000Hz
724 // FIX handle wrap-around
725 if (mProcessing[0].mRTPTimeStamp + 20*(48000/1000) >= now) {
726 TimeDuration t = TimeStamp::Now() - mProcessing[0].mTimeStamp;
727 // Wrap-around?
728 int64_t delta = t.ToMilliseconds() + (now - mProcessing[0].mRTPTimeStamp)/(48000/1000);
729 LogTime(AsyncLatencyLogger::AudioRecvRTP, ((uint64_t) this), delta);
730 break;
731 }
732 mProcessing.RemoveElementAt(0);
733 }
734 }
735 }
736 }
737 #endif
738 CSFLogDebug(logTag,"%s GetAudioFrame:Got samples: length %d ",__FUNCTION__,
739 lengthSamples);
740 return kMediaConduitNoError;
741 }
743 // Transport Layer Callbacks
744 MediaConduitErrorCode
745 WebrtcAudioConduit::ReceivedRTPPacket(const void *data, int len)
746 {
747 CSFLogDebug(logTag, "%s : channel %d", __FUNCTION__, mChannel);
749 if(mEngineReceiving)
750 {
751 #ifdef MOZILLA_INTERNAL_API
752 if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
753 // timestamp is at 32 bits in ([1])
754 struct Processing insert = { TimeStamp::Now(),
755 ntohl(static_cast<const uint32_t *>(data)[1]) };
756 mProcessing.AppendElement(insert);
757 }
758 #endif
760 if(mPtrVoENetwork->ReceivedRTPPacket(mChannel,data,len) == -1)
761 {
762 int error = mPtrVoEBase->LastError();
763 CSFLogError(logTag, "%s RTP Processing Error %d", __FUNCTION__, error);
764 if(error == VE_RTP_RTCP_MODULE_ERROR)
765 {
766 return kMediaConduitRTPRTCPModuleError;
767 }
768 return kMediaConduitUnknownError;
769 }
770 } else {
771 CSFLogError(logTag, "Error: %s when not receiving", __FUNCTION__);
772 return kMediaConduitSessionNotInited;
773 }
775 return kMediaConduitNoError;
776 }
778 MediaConduitErrorCode
779 WebrtcAudioConduit::ReceivedRTCPPacket(const void *data, int len)
780 {
781 CSFLogDebug(logTag, "%s : channel %d",__FUNCTION__, mChannel);
783 if(mEngineTransmitting)
784 {
785 if(mPtrVoENetwork->ReceivedRTCPPacket(mChannel, data, len) == -1)
786 {
787 int error = mPtrVoEBase->LastError();
788 CSFLogError(logTag, "%s RTCP Processing Error %d", __FUNCTION__, error);
789 if(error == VE_RTP_RTCP_MODULE_ERROR)
790 {
791 return kMediaConduitRTPRTCPModuleError;
792 }
793 return kMediaConduitUnknownError;
794 }
795 } else {
796 CSFLogError(logTag, "Error: %s when not receiving", __FUNCTION__);
797 return kMediaConduitSessionNotInited;
798 }
799 return kMediaConduitNoError;
800 }
802 //WebRTC::RTP Callback Implementation
803 int WebrtcAudioConduit::SendPacket(int channel, const void* data, int len)
804 {
805 CSFLogDebug(logTag, "%s : channel %d %s", __FUNCTION__, channel,
806 (mEngineReceiving && mOtherDirection) ? "(using mOtherDirection)" : "");
808 if (mEngineReceiving)
809 {
810 if (mOtherDirection)
811 {
812 return mOtherDirection->SendPacket(channel, data, len);
813 }
814 CSFLogDebug(logTag, "%s : Asked to send RTP without an RTP sender on channel %d",
815 __FUNCTION__, channel);
816 return -1;
817 } else {
818 #ifdef MOZILLA_INTERNAL_API
819 if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
820 if (mProcessing.Length() > 0) {
821 TimeStamp started = mProcessing[0].mTimeStamp;
822 mProcessing.RemoveElementAt(0);
823 mProcessing.RemoveElementAt(0); // 20ms packetization! Could automate this by watching sizes
824 TimeDuration t = TimeStamp::Now() - started;
825 int64_t delta = t.ToMilliseconds();
826 LogTime(AsyncLatencyLogger::AudioSendRTP, ((uint64_t) this), delta);
827 }
828 }
829 #endif
830 if(mTransport && (mTransport->SendRtpPacket(data, len) == NS_OK))
831 {
832 CSFLogDebug(logTag, "%s Sent RTP Packet ", __FUNCTION__);
833 return len;
834 } else {
835 CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);
836 return -1;
837 }
838 }
839 }
841 int WebrtcAudioConduit::SendRTCPPacket(int channel, const void* data, int len)
842 {
843 CSFLogDebug(logTag, "%s : channel %d", __FUNCTION__, channel);
845 if (mEngineTransmitting)
846 {
847 if (mOtherDirection)
848 {
849 return mOtherDirection->SendRTCPPacket(channel, data, len);
850 }
851 }
853 // We come here if we have only one pipeline/conduit setup,
854 // such as for unidirectional streams.
855 // We also end up here if we are receiving
856 if(mTransport && mTransport->SendRtcpPacket(data, len) == NS_OK)
857 {
858 CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
859 return len;
860 } else {
861 CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
862 return -1;
863 }
864 }
866 /**
867 * Converts between CodecConfig to WebRTC Codec Structure.
868 */
870 bool
871 WebrtcAudioConduit::CodecConfigToWebRTCCodec(const AudioCodecConfig* codecInfo,
872 webrtc::CodecInst& cinst)
873 {
874 const unsigned int plNameLength = codecInfo->mName.length()+1;
875 memset(&cinst, 0, sizeof(webrtc::CodecInst));
876 if(sizeof(cinst.plname) < plNameLength)
877 {
878 CSFLogError(logTag, "%s Payload name buffer capacity mismatch ",
879 __FUNCTION__);
880 return false;
881 }
882 memcpy(cinst.plname, codecInfo->mName.c_str(),codecInfo->mName.length());
883 cinst.plname[plNameLength]='\0';
884 cinst.pltype = codecInfo->mType;
885 cinst.rate = codecInfo->mRate;
886 cinst.pacsize = codecInfo->mPacSize;
887 cinst.plfreq = codecInfo->mFreq;
888 cinst.channels = codecInfo->mChannels;
889 return true;
890 }
892 /**
893 * Supported Sampling Frequncies.
894 */
895 bool
896 WebrtcAudioConduit::IsSamplingFreqSupported(int freq) const
897 {
898 if(GetNum10msSamplesForFrequency(freq))
899 {
900 return true;
901 } else {
902 return false;
903 }
904 }
906 /* Return block-length of 10 ms audio frame in number of samples */
907 unsigned int
908 WebrtcAudioConduit::GetNum10msSamplesForFrequency(int samplingFreqHz) const
909 {
910 switch(samplingFreqHz)
911 {
912 case 16000: return 160; //160 samples
913 case 32000: return 320; //320 samples
914 case 44100: return 441; //441 samples
915 case 48000: return 480; //480 samples
916 default: return 0; // invalid or unsupported
917 }
918 }
920 //Copy the codec passed into Conduit's database
921 bool
922 WebrtcAudioConduit::CopyCodecToDB(const AudioCodecConfig* codecInfo)
923 {
925 AudioCodecConfig* cdcConfig = new AudioCodecConfig(codecInfo->mType,
926 codecInfo->mName,
927 codecInfo->mFreq,
928 codecInfo->mPacSize,
929 codecInfo->mChannels,
930 codecInfo->mRate,
931 codecInfo->mLoadManager);
932 mRecvCodecList.push_back(cdcConfig);
933 return true;
934 }
936 /**
937 * Checks if 2 codec structs are same
938 */
939 bool
940 WebrtcAudioConduit::CheckCodecsForMatch(const AudioCodecConfig* curCodecConfig,
941 const AudioCodecConfig* codecInfo) const
942 {
943 if(!curCodecConfig)
944 {
945 return false;
946 }
948 if(curCodecConfig->mType == codecInfo->mType &&
949 (curCodecConfig->mName.compare(codecInfo->mName) == 0) &&
950 curCodecConfig->mFreq == codecInfo->mFreq &&
951 curCodecConfig->mPacSize == codecInfo->mPacSize &&
952 curCodecConfig->mChannels == codecInfo->mChannels &&
953 curCodecConfig->mRate == codecInfo->mRate)
954 {
955 return true;
956 }
958 return false;
959 }
961 /**
962 * Checks if the codec is already in Conduit's database
963 */
964 bool
965 WebrtcAudioConduit::CheckCodecForMatch(const AudioCodecConfig* codecInfo) const
966 {
967 //the db should have atleast one codec
968 for(std::vector<AudioCodecConfig*>::size_type i=0;i < mRecvCodecList.size();i++)
969 {
970 if(CheckCodecsForMatch(mRecvCodecList[i],codecInfo))
971 {
972 //match
973 return true;
974 }
975 }
976 //no match or empty local db
977 return false;
978 }
981 /**
982 * Perform validation on the codecConfig to be applied.
983 * Verifies if the codec is already applied.
984 */
985 MediaConduitErrorCode
986 WebrtcAudioConduit::ValidateCodecConfig(const AudioCodecConfig* codecInfo,
987 bool send) const
988 {
989 bool codecAppliedAlready = false;
991 if(!codecInfo)
992 {
993 CSFLogError(logTag, "%s Null CodecConfig ", __FUNCTION__);
994 return kMediaConduitMalformedArgument;
995 }
997 if((codecInfo->mName.empty()) ||
998 (codecInfo->mName.length() >= CODEC_PLNAME_SIZE))
999 {
1000 CSFLogError(logTag, "%s Invalid Payload Name Length ", __FUNCTION__);
1001 return kMediaConduitMalformedArgument;
1002 }
1004 //Only mono or stereo channels supported
1005 if( (codecInfo->mChannels != 1) && (codecInfo->mChannels != 2))
1006 {
1007 CSFLogError(logTag, "%s Channel Unsupported ", __FUNCTION__);
1008 return kMediaConduitMalformedArgument;
1009 }
1011 //check if we have the same codec already applied
1012 if(send)
1013 {
1014 codecAppliedAlready = CheckCodecsForMatch(mCurSendCodecConfig,codecInfo);
1015 } else {
1016 codecAppliedAlready = CheckCodecForMatch(codecInfo);
1017 }
1019 if(codecAppliedAlready)
1020 {
1021 CSFLogDebug(logTag, "%s Codec %s Already Applied ", __FUNCTION__, codecInfo->mName.c_str());
1022 return kMediaConduitCodecInUse;
1023 }
1024 return kMediaConduitNoError;
1025 }
1027 void
1028 WebrtcAudioConduit::DumpCodecDB() const
1029 {
1030 for(std::vector<AudioCodecConfig*>::size_type i=0;i < mRecvCodecList.size();i++)
1031 {
1032 CSFLogDebug(logTag,"Payload Name: %s", mRecvCodecList[i]->mName.c_str());
1033 CSFLogDebug(logTag,"Payload Type: %d", mRecvCodecList[i]->mType);
1034 CSFLogDebug(logTag,"Payload Frequency: %d", mRecvCodecList[i]->mFreq);
1035 CSFLogDebug(logTag,"Payload PacketSize: %d", mRecvCodecList[i]->mPacSize);
1036 CSFLogDebug(logTag,"Payload Channels: %d", mRecvCodecList[i]->mChannels);
1037 CSFLogDebug(logTag,"Payload Sampling Rate: %d", mRecvCodecList[i]->mRate);
1038 }
1039 }
1040 }// end namespace
