1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 #include "content/renderer/media/rtc_peer_connection_handler.h"
11 #include "base/command_line.h"
12 #include "base/lazy_instance.h"
13 #include "base/location.h"
14 #include "base/logging.h"
15 #include "base/memory/scoped_ptr.h"
16 #include "base/metrics/histogram.h"
17 #include "base/stl_util.h"
18 #include "base/strings/utf_string_conversions.h"
19 #include "base/thread_task_runner_handle.h"
20 #include "base/trace_event/trace_event.h"
21 #include "content/public/common/content_switches.h"
22 #include "content/renderer/media/media_stream_track.h"
23 #include "content/renderer/media/peer_connection_tracker.h"
24 #include "content/renderer/media/remote_media_stream_impl.h"
25 #include "content/renderer/media/rtc_data_channel_handler.h"
26 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
27 #include "content/renderer/media/rtc_media_constraints.h"
28 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
29 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
30 #include "content/renderer/media/webrtc_audio_capturer.h"
31 #include "content/renderer/media/webrtc_audio_device_impl.h"
32 #include "content/renderer/media/webrtc_uma_histograms.h"
33 #include "content/renderer/render_thread_impl.h"
34 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
35 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
36 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
37 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
38 #include "third_party/WebKit/public/platform/WebRTCOfferOptions.h"
39 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
40 #include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
41 #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
42 #include "third_party/WebKit/public/platform/WebURL.h"
44 using webrtc::DataChannelInterface
;
45 using webrtc::IceCandidateInterface
;
46 using webrtc::MediaStreamInterface
;
47 using webrtc::PeerConnectionInterface
;
48 using webrtc::PeerConnectionObserver
;
49 using webrtc::StatsReport
;
50 using webrtc::StatsReports
;
55 // Converter functions from libjingle types to WebKit types.
56 blink::WebRTCPeerConnectionHandlerClient::ICEGatheringState
57 GetWebKitIceGatheringState(
58 webrtc::PeerConnectionInterface::IceGatheringState state
) {
59 using blink::WebRTCPeerConnectionHandlerClient
;
61 case webrtc::PeerConnectionInterface::kIceGatheringNew
:
62 return WebRTCPeerConnectionHandlerClient::ICEGatheringStateNew
;
63 case webrtc::PeerConnectionInterface::kIceGatheringGathering
:
64 return WebRTCPeerConnectionHandlerClient::ICEGatheringStateGathering
;
65 case webrtc::PeerConnectionInterface::kIceGatheringComplete
:
66 return WebRTCPeerConnectionHandlerClient::ICEGatheringStateComplete
;
69 return WebRTCPeerConnectionHandlerClient::ICEGatheringStateNew
;
73 blink::WebRTCPeerConnectionHandlerClient::ICEConnectionState
74 GetWebKitIceConnectionState(
75 webrtc::PeerConnectionInterface::IceConnectionState ice_state
) {
76 using blink::WebRTCPeerConnectionHandlerClient
;
78 case webrtc::PeerConnectionInterface::kIceConnectionNew
:
79 return WebRTCPeerConnectionHandlerClient::ICEConnectionStateStarting
;
80 case webrtc::PeerConnectionInterface::kIceConnectionChecking
:
81 return WebRTCPeerConnectionHandlerClient::ICEConnectionStateChecking
;
82 case webrtc::PeerConnectionInterface::kIceConnectionConnected
:
83 return WebRTCPeerConnectionHandlerClient::ICEConnectionStateConnected
;
84 case webrtc::PeerConnectionInterface::kIceConnectionCompleted
:
85 return WebRTCPeerConnectionHandlerClient::ICEConnectionStateCompleted
;
86 case webrtc::PeerConnectionInterface::kIceConnectionFailed
:
87 return WebRTCPeerConnectionHandlerClient::ICEConnectionStateFailed
;
88 case webrtc::PeerConnectionInterface::kIceConnectionDisconnected
:
89 return WebRTCPeerConnectionHandlerClient::ICEConnectionStateDisconnected
;
90 case webrtc::PeerConnectionInterface::kIceConnectionClosed
:
91 return WebRTCPeerConnectionHandlerClient::ICEConnectionStateClosed
;
94 return WebRTCPeerConnectionHandlerClient::ICEConnectionStateClosed
;
98 blink::WebRTCPeerConnectionHandlerClient::SignalingState
99 GetWebKitSignalingState(webrtc::PeerConnectionInterface::SignalingState state
) {
100 using blink::WebRTCPeerConnectionHandlerClient
;
102 case webrtc::PeerConnectionInterface::kStable
:
103 return WebRTCPeerConnectionHandlerClient::SignalingStateStable
;
104 case webrtc::PeerConnectionInterface::kHaveLocalOffer
:
105 return WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalOffer
;
106 case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer
:
107 return WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalPrAnswer
;
108 case webrtc::PeerConnectionInterface::kHaveRemoteOffer
:
109 return WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemoteOffer
;
110 case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer
:
112 WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemotePrAnswer
;
113 case webrtc::PeerConnectionInterface::kClosed
:
114 return WebRTCPeerConnectionHandlerClient::SignalingStateClosed
;
117 return WebRTCPeerConnectionHandlerClient::SignalingStateClosed
;
121 blink::WebRTCSessionDescription
CreateWebKitSessionDescription(
122 const std::string
& sdp
, const std::string
& type
) {
123 blink::WebRTCSessionDescription description
;
124 description
.initialize(base::UTF8ToUTF16(type
), base::UTF8ToUTF16(sdp
));
128 blink::WebRTCSessionDescription
129 CreateWebKitSessionDescription(
130 const webrtc::SessionDescriptionInterface
* native_desc
) {
132 LOG(ERROR
) << "Native session description is null.";
133 return blink::WebRTCSessionDescription();
137 if (!native_desc
->ToString(&sdp
)) {
138 LOG(ERROR
) << "Failed to get SDP string of native session description.";
139 return blink::WebRTCSessionDescription();
142 return CreateWebKitSessionDescription(sdp
, native_desc
->type());
145 void RunClosureWithTrace(const base::Closure
& closure
,
146 const char* trace_event_name
) {
147 TRACE_EVENT0("webrtc", trace_event_name
);
151 void RunSynchronousClosure(const base::Closure
& closure
,
152 const char* trace_event_name
,
153 base::WaitableEvent
* event
) {
155 TRACE_EVENT0("webrtc", trace_event_name
);
161 void GetSdpAndTypeFromSessionDescription(
162 const base::Callback
<const webrtc::SessionDescriptionInterface
*()>&
163 description_callback
,
164 std::string
* sdp
, std::string
* type
) {
165 const webrtc::SessionDescriptionInterface
* description
=
166 description_callback
.Run();
168 description
->ToString(sdp
);
169 *type
= description
->type();
173 // Converter functions from WebKit types to WebRTC types.
175 void GetNativeRtcConfiguration(
176 const blink::WebRTCConfiguration
& blink_config
,
177 webrtc::PeerConnectionInterface::RTCConfiguration
* webrtc_config
) {
178 if (blink_config
.isNull() || !webrtc_config
)
180 for (size_t i
= 0; i
< blink_config
.numberOfServers(); ++i
) {
181 webrtc::PeerConnectionInterface::IceServer server
;
182 const blink::WebRTCICEServer
& webkit_server
=
183 blink_config
.server(i
);
185 base::UTF16ToUTF8(base::StringPiece16(webkit_server
.username()));
187 base::UTF16ToUTF8(base::StringPiece16(webkit_server
.credential()));
188 server
.uri
= webkit_server
.uri().spec();
189 webrtc_config
->servers
.push_back(server
);
192 switch (blink_config
.iceTransports()) {
193 case blink::WebRTCIceTransportsNone
:
194 webrtc_config
->type
= webrtc::PeerConnectionInterface::kNone
;
196 case blink::WebRTCIceTransportsRelay
:
197 webrtc_config
->type
= webrtc::PeerConnectionInterface::kRelay
;
199 case blink::WebRTCIceTransportsAll
:
200 webrtc_config
->type
= webrtc::PeerConnectionInterface::kAll
;
206 switch (blink_config
.bundlePolicy()) {
207 case blink::WebRTCBundlePolicyBalanced
:
208 webrtc_config
->bundle_policy
=
209 webrtc::PeerConnectionInterface::kBundlePolicyBalanced
;
211 case blink::WebRTCBundlePolicyMaxBundle
:
212 webrtc_config
->bundle_policy
=
213 webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle
;
215 case blink::WebRTCBundlePolicyMaxCompat
:
216 webrtc_config
->bundle_policy
=
217 webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat
;
223 switch (blink_config
.rtcpMuxPolicy()) {
224 case blink::WebRTCRtcpMuxPolicyNegotiate
:
225 webrtc_config
->rtcp_mux_policy
=
226 webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate
;
228 case blink::WebRTCRtcpMuxPolicyRequire
:
229 webrtc_config
->rtcp_mux_policy
=
230 webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire
;
237 class SessionDescriptionRequestTracker
{
239 SessionDescriptionRequestTracker(
240 const base::WeakPtr
<RTCPeerConnectionHandler
>& handler
,
241 const base::WeakPtr
<PeerConnectionTracker
>& tracker
,
242 PeerConnectionTracker::Action action
)
243 : handler_(handler
), tracker_(tracker
), action_(action
) {}
245 void TrackOnSuccess(const webrtc::SessionDescriptionInterface
* desc
) {
246 DCHECK(thread_checker_
.CalledOnValidThread());
247 if (tracker_
&& handler_
) {
250 desc
->ToString(&value
);
251 value
= "type: " + desc
->type() + ", sdp: " + value
;
253 tracker_
->TrackSessionDescriptionCallback(
254 handler_
.get(), action_
, "OnSuccess", value
);
258 void TrackOnFailure(const std::string
& error
) {
259 DCHECK(thread_checker_
.CalledOnValidThread());
260 if (handler_
&& tracker_
) {
261 tracker_
->TrackSessionDescriptionCallback(
262 handler_
.get(), action_
, "OnFailure", error
);
267 const base::WeakPtr
<RTCPeerConnectionHandler
> handler_
;
268 const base::WeakPtr
<PeerConnectionTracker
> tracker_
;
269 PeerConnectionTracker::Action action_
;
270 base::ThreadChecker thread_checker_
;
273 // Class mapping responses from calls to libjingle CreateOffer/Answer and
274 // the blink::WebRTCSessionDescriptionRequest.
275 class CreateSessionDescriptionRequest
276 : public webrtc::CreateSessionDescriptionObserver
{
278 explicit CreateSessionDescriptionRequest(
279 const scoped_refptr
<base::SingleThreadTaskRunner
>& main_thread
,
280 const blink::WebRTCSessionDescriptionRequest
& request
,
281 const base::WeakPtr
<RTCPeerConnectionHandler
>& handler
,
282 const base::WeakPtr
<PeerConnectionTracker
>& tracker
,
283 PeerConnectionTracker::Action action
)
284 : main_thread_(main_thread
),
285 webkit_request_(request
),
286 tracker_(handler
, tracker
, action
) {
289 void OnSuccess(webrtc::SessionDescriptionInterface
* desc
) override
{
290 if (!main_thread_
->BelongsToCurrentThread()) {
291 main_thread_
->PostTask(FROM_HERE
,
292 base::Bind(&CreateSessionDescriptionRequest::OnSuccess
, this, desc
));
296 tracker_
.TrackOnSuccess(desc
);
297 webkit_request_
.requestSucceeded(CreateWebKitSessionDescription(desc
));
298 webkit_request_
.reset();
301 void OnFailure(const std::string
& error
) override
{
302 if (!main_thread_
->BelongsToCurrentThread()) {
303 main_thread_
->PostTask(FROM_HERE
,
304 base::Bind(&CreateSessionDescriptionRequest::OnFailure
, this, error
));
308 tracker_
.TrackOnFailure(error
);
309 webkit_request_
.requestFailed(base::UTF8ToUTF16(error
));
310 webkit_request_
.reset();
314 ~CreateSessionDescriptionRequest() override
{
315 CHECK(webkit_request_
.isNull());
318 const scoped_refptr
<base::SingleThreadTaskRunner
> main_thread_
;
319 blink::WebRTCSessionDescriptionRequest webkit_request_
;
320 SessionDescriptionRequestTracker tracker_
;
323 // Class mapping responses from calls to libjingle
324 // SetLocalDescription/SetRemoteDescription and a blink::WebRTCVoidRequest.
325 class SetSessionDescriptionRequest
326 : public webrtc::SetSessionDescriptionObserver
{
328 explicit SetSessionDescriptionRequest(
329 const scoped_refptr
<base::SingleThreadTaskRunner
>& main_thread
,
330 const blink::WebRTCVoidRequest
& request
,
331 const base::WeakPtr
<RTCPeerConnectionHandler
>& handler
,
332 const base::WeakPtr
<PeerConnectionTracker
>& tracker
,
333 PeerConnectionTracker::Action action
)
334 : main_thread_(main_thread
),
335 webkit_request_(request
),
336 tracker_(handler
, tracker
, action
) {
339 void OnSuccess() override
{
340 if (!main_thread_
->BelongsToCurrentThread()) {
341 main_thread_
->PostTask(FROM_HERE
,
342 base::Bind(&SetSessionDescriptionRequest::OnSuccess
, this));
345 tracker_
.TrackOnSuccess(NULL
);
346 webkit_request_
.requestSucceeded();
347 webkit_request_
.reset();
349 void OnFailure(const std::string
& error
) override
{
350 if (!main_thread_
->BelongsToCurrentThread()) {
351 main_thread_
->PostTask(FROM_HERE
,
352 base::Bind(&SetSessionDescriptionRequest::OnFailure
, this, error
));
355 tracker_
.TrackOnFailure(error
);
356 webkit_request_
.requestFailed(base::UTF8ToUTF16(error
));
357 webkit_request_
.reset();
361 ~SetSessionDescriptionRequest() override
{
362 DCHECK(webkit_request_
.isNull());
366 const scoped_refptr
<base::SingleThreadTaskRunner
> main_thread_
;
367 blink::WebRTCVoidRequest webkit_request_
;
368 SessionDescriptionRequestTracker tracker_
;
371 // Class mapping responses from calls to libjingle
372 // GetStats into a blink::WebRTCStatsCallback.
373 class StatsResponse
: public webrtc::StatsObserver
{
375 explicit StatsResponse(const scoped_refptr
<LocalRTCStatsRequest
>& request
)
376 : request_(request
.get()),
377 main_thread_(base::ThreadTaskRunnerHandle::Get()) {
378 // Measure the overall time it takes to satisfy a getStats request.
379 TRACE_EVENT_ASYNC_BEGIN0("webrtc", "getStats_Native", this);
380 signaling_thread_checker_
.DetachFromThread();
383 void OnComplete(const StatsReports
& reports
) override
{
384 DCHECK(signaling_thread_checker_
.CalledOnValidThread());
385 TRACE_EVENT0("webrtc", "StatsResponse::OnComplete");
386 // We can't use webkit objects directly since they use a single threaded
388 std::vector
<Report
*>* report_copies
= new std::vector
<Report
*>();
389 report_copies
->reserve(reports
.size());
390 for (auto* r
: reports
)
391 report_copies
->push_back(new Report(r
));
393 main_thread_
->PostTaskAndReply(FROM_HERE
,
394 base::Bind(&StatsResponse::DeliverCallback
, this,
395 base::Unretained(report_copies
)),
396 base::Bind(&StatsResponse::DeleteReports
,
397 base::Unretained(report_copies
)));
402 Report(const StatsReport
* report
)
403 : thread_checker(), id(report
->id()->ToString()),
404 type(report
->TypeToString()), timestamp(report
->timestamp()),
405 values(report
->values()) {
409 // Since the values vector holds pointers to const objects that are bound
410 // to the signaling thread, they must be released on the same thread.
411 DCHECK(thread_checker
.CalledOnValidThread());
414 const base::ThreadChecker thread_checker
;
415 const std::string id
, type
;
416 const double timestamp
;
417 const StatsReport::Values values
;
420 static void DeleteReports(std::vector
<Report
*>* reports
) {
421 TRACE_EVENT0("webrtc", "StatsResponse::DeleteReports");
422 for (auto* p
: *reports
)
427 void DeliverCallback(const std::vector
<Report
*>* reports
) {
428 DCHECK(main_thread_
->BelongsToCurrentThread());
429 TRACE_EVENT0("webrtc", "StatsResponse::DeliverCallback");
431 rtc::scoped_refptr
<LocalRTCStatsResponse
> response(
432 request_
->createResponse().get());
433 for (const auto* report
: *reports
) {
434 if (report
->values
.size() > 0)
435 AddReport(response
.get(), *report
);
438 // Record the getStats operation as done before calling into Blink so that
439 // we don't skew the perf measurements of the native code with whatever the
440 // callback might be doing.
441 TRACE_EVENT_ASYNC_END0("webrtc", "getStats_Native", this);
442 request_
->requestSucceeded(response
);
443 request_
= nullptr; // must be freed on the main thread.
446 void AddReport(LocalRTCStatsResponse
* response
, const Report
& report
) {
447 int idx
= response
->addReport(blink::WebString::fromUTF8(report
.id
),
448 blink::WebString::fromUTF8(report
.type
),
450 blink::WebString name
, value_str
;
451 for (const auto& value
: report
.values
) {
452 const StatsReport::ValuePtr
& v
= value
.second
;
453 name
= blink::WebString::fromUTF8(value
.second
->display_name());
455 if (v
->type() == StatsReport::Value::kString
)
456 value_str
= blink::WebString::fromUTF8(v
->string_val());
457 if (v
->type() == StatsReport::Value::kStaticString
)
458 value_str
= blink::WebString::fromUTF8(v
->static_string_val());
460 value_str
= blink::WebString::fromUTF8(v
->ToString());
462 response
->addStatistic(idx
, name
, value_str
);
466 rtc::scoped_refptr
<LocalRTCStatsRequest
> request_
;
467 const scoped_refptr
<base::SingleThreadTaskRunner
> main_thread_
;
468 base::ThreadChecker signaling_thread_checker_
;
471 void GetStatsOnSignalingThread(
472 const scoped_refptr
<webrtc::PeerConnectionInterface
>& pc
,
473 webrtc::PeerConnectionInterface::StatsOutputLevel level
,
474 const scoped_refptr
<webrtc::StatsObserver
>& observer
,
475 const std::string track_id
, blink::WebMediaStreamSource::Type track_type
) {
476 TRACE_EVENT0("webrtc", "GetStatsOnSignalingThread");
478 scoped_refptr
<webrtc::MediaStreamTrackInterface
> track
;
479 if (!track_id
.empty()) {
480 if (track_type
== blink::WebMediaStreamSource::TypeAudio
) {
481 track
= pc
->local_streams()->FindAudioTrack(track_id
);
483 track
= pc
->remote_streams()->FindAudioTrack(track_id
);
485 DCHECK_EQ(blink::WebMediaStreamSource::TypeVideo
, track_type
);
486 track
= pc
->local_streams()->FindVideoTrack(track_id
);
488 track
= pc
->remote_streams()->FindVideoTrack(track_id
);
492 DVLOG(1) << "GetStats: Track not found.";
493 observer
->OnComplete(StatsReports());
498 if (!pc
->GetStats(observer
.get(), track
.get(), level
)) {
499 DVLOG(1) << "GetStats failed.";
500 observer
->OnComplete(StatsReports());
504 class PeerConnectionUMAObserver
: public webrtc::UMAObserver
{
506 PeerConnectionUMAObserver() {}
507 ~PeerConnectionUMAObserver() override
{}
509 void IncrementCounter(
510 webrtc::PeerConnectionUMAMetricsCounter counter
) override
{
511 // Runs on libjingle's signaling thread.
512 UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
517 void AddHistogramSample(webrtc::PeerConnectionUMAMetricsName type
,
518 int value
) override
{
519 // Runs on libjingle's signaling thread.
521 case webrtc::kTimeToConnect
:
522 UMA_HISTOGRAM_MEDIUM_TIMES(
523 "WebRTC.PeerConnection.TimeToConnect",
524 base::TimeDelta::FromMilliseconds(value
));
526 case webrtc::kNetworkInterfaces_IPv4
:
527 UMA_HISTOGRAM_COUNTS_100("WebRTC.PeerConnection.IPv4Interfaces",
530 case webrtc::kNetworkInterfaces_IPv6
:
531 UMA_HISTOGRAM_COUNTS_100("WebRTC.PeerConnection.IPv6Interfaces",
540 base::LazyInstance
<std::set
<RTCPeerConnectionHandler
*> >::Leaky
541 g_peer_connection_handlers
= LAZY_INSTANCE_INITIALIZER
;
545 // Implementation of LocalRTCStatsRequest.
546 LocalRTCStatsRequest::LocalRTCStatsRequest(blink::WebRTCStatsRequest impl
)
550 LocalRTCStatsRequest::LocalRTCStatsRequest() {}
551 LocalRTCStatsRequest::~LocalRTCStatsRequest() {}
553 bool LocalRTCStatsRequest::hasSelector() const {
554 return impl_
.hasSelector();
557 blink::WebMediaStreamTrack
LocalRTCStatsRequest::component() const {
558 return impl_
.component();
561 scoped_refptr
<LocalRTCStatsResponse
> LocalRTCStatsRequest::createResponse() {
562 return scoped_refptr
<LocalRTCStatsResponse
>(
563 new rtc::RefCountedObject
<LocalRTCStatsResponse
>(impl_
.createResponse()));
566 void LocalRTCStatsRequest::requestSucceeded(
567 const LocalRTCStatsResponse
* response
) {
568 impl_
.requestSucceeded(response
->webKitStatsResponse());
571 // Implementation of LocalRTCStatsResponse.
572 blink::WebRTCStatsResponse
LocalRTCStatsResponse::webKitStatsResponse() const {
576 size_t LocalRTCStatsResponse::addReport(blink::WebString type
,
579 return impl_
.addReport(type
, id
, timestamp
);
582 void LocalRTCStatsResponse::addStatistic(size_t report
,
583 blink::WebString name
,
584 blink::WebString value
) {
585 impl_
.addStatistic(report
, name
, value
);
588 // Receives notifications from a PeerConnection object about state changes,
589 // track addition/removal etc. The callbacks we receive here come on the
590 // signaling thread, so this class takes care of delivering them to an
591 // RTCPeerConnectionHandler instance on the main thread.
592 // In order to do safe PostTask-ing, the class is reference counted and
593 // checks for the existence of the RTCPeerConnectionHandler instance before
594 // delivering callbacks on the main thread.
595 class RTCPeerConnectionHandler::Observer
596 : public base::RefCountedThreadSafe
<RTCPeerConnectionHandler::Observer
>,
597 public PeerConnectionObserver
{
599 Observer(const base::WeakPtr
<RTCPeerConnectionHandler
>& handler
)
600 : handler_(handler
), main_thread_(base::ThreadTaskRunnerHandle::Get()) {}
603 friend class base::RefCountedThreadSafe
<RTCPeerConnectionHandler::Observer
>;
604 virtual ~Observer() {}
606 void OnSignalingChange(
607 PeerConnectionInterface::SignalingState new_state
) override
{
608 if (!main_thread_
->BelongsToCurrentThread()) {
609 main_thread_
->PostTask(FROM_HERE
,
610 base::Bind(&RTCPeerConnectionHandler::Observer::OnSignalingChange
,
612 } else if (handler_
) {
613 handler_
->OnSignalingChange(new_state
);
617 void OnAddStream(MediaStreamInterface
* stream
) override
{
619 scoped_ptr
<RemoteMediaStreamImpl
> remote_stream(
620 new RemoteMediaStreamImpl(main_thread_
, stream
));
622 // The webkit object owned by RemoteMediaStreamImpl, will be initialized
623 // asynchronously and the posted task will execude after that initialization
625 main_thread_
->PostTask(FROM_HERE
,
626 base::Bind(&RTCPeerConnectionHandler::Observer::OnAddStreamImpl
,
627 this, base::Passed(&remote_stream
)));
630 void OnRemoveStream(MediaStreamInterface
* stream
) override
{
631 main_thread_
->PostTask(FROM_HERE
,
632 base::Bind(&RTCPeerConnectionHandler::Observer::OnRemoveStreamImpl
,
633 this, make_scoped_refptr(stream
)));
636 void OnDataChannel(DataChannelInterface
* data_channel
) override
{
637 scoped_ptr
<RtcDataChannelHandler
> handler(
638 new RtcDataChannelHandler(main_thread_
, data_channel
));
639 main_thread_
->PostTask(FROM_HERE
,
640 base::Bind(&RTCPeerConnectionHandler::Observer::OnDataChannelImpl
,
641 this, base::Passed(&handler
)));
644 void OnRenegotiationNeeded() override
{
645 if (!main_thread_
->BelongsToCurrentThread()) {
646 main_thread_
->PostTask(FROM_HERE
,
647 base::Bind(&RTCPeerConnectionHandler::Observer::OnRenegotiationNeeded
,
649 } else if (handler_
) {
650 handler_
->OnRenegotiationNeeded();
654 void OnIceConnectionChange(
655 PeerConnectionInterface::IceConnectionState new_state
) override
{
656 if (!main_thread_
->BelongsToCurrentThread()) {
657 main_thread_
->PostTask(FROM_HERE
,
659 &RTCPeerConnectionHandler::Observer::OnIceConnectionChange
, this,
661 } else if (handler_
) {
662 handler_
->OnIceConnectionChange(new_state
);
666 void OnIceGatheringChange(
667 PeerConnectionInterface::IceGatheringState new_state
) override
{
668 if (!main_thread_
->BelongsToCurrentThread()) {
669 main_thread_
->PostTask(FROM_HERE
,
670 base::Bind(&RTCPeerConnectionHandler::Observer::OnIceGatheringChange
,
672 } else if (handler_
) {
673 handler_
->OnIceGatheringChange(new_state
);
677 void OnIceCandidate(const IceCandidateInterface
* candidate
) override
{
679 if (!candidate
->ToString(&sdp
)) {
680 NOTREACHED() << "OnIceCandidate: Could not get SDP string.";
684 main_thread_
->PostTask(FROM_HERE
,
685 base::Bind(&RTCPeerConnectionHandler::Observer::OnIceCandidateImpl
,
686 this, sdp
, candidate
->sdp_mid(), candidate
->sdp_mline_index(),
687 candidate
->candidate().component(),
688 candidate
->candidate().address().family()));
691 void OnAddStreamImpl(scoped_ptr
<RemoteMediaStreamImpl
> stream
) {
692 DCHECK(stream
->webkit_stream().extraData()) << "Initialization not done";
694 handler_
->OnAddStream(stream
.Pass());
697 void OnRemoveStreamImpl(const scoped_refptr
<MediaStreamInterface
>& stream
) {
699 handler_
->OnRemoveStream(stream
);
702 void OnDataChannelImpl(scoped_ptr
<RtcDataChannelHandler
> handler
) {
704 handler_
->OnDataChannel(handler
.Pass());
707 void OnIceCandidateImpl(const std::string
& sdp
, const std::string
& sdp_mid
,
708 int sdp_mline_index
, int component
, int address_family
) {
710 handler_
->OnIceCandidate(sdp
, sdp_mid
, sdp_mline_index
, component
,
716 const base::WeakPtr
<RTCPeerConnectionHandler
> handler_
;
717 const scoped_refptr
<base::SingleThreadTaskRunner
> main_thread_
;
720 RTCPeerConnectionHandler::RTCPeerConnectionHandler(
721 blink::WebRTCPeerConnectionHandlerClient
* client
,
722 PeerConnectionDependencyFactory
* dependency_factory
)
724 dependency_factory_(dependency_factory
),
726 num_data_channels_created_(0),
727 num_local_candidates_ipv4_(0),
728 num_local_candidates_ipv6_(0),
729 weak_factory_(this) {
730 g_peer_connection_handlers
.Get().insert(this);
733 RTCPeerConnectionHandler::~RTCPeerConnectionHandler() {
734 DCHECK(thread_checker_
.CalledOnValidThread());
738 g_peer_connection_handlers
.Get().erase(this);
739 if (peer_connection_tracker_
)
740 peer_connection_tracker_
->UnregisterPeerConnection(this);
741 STLDeleteValues(&remote_streams_
);
743 UMA_HISTOGRAM_COUNTS_10000(
744 "WebRTC.NumDataChannelsPerPeerConnection", num_data_channels_created_
);
748 void RTCPeerConnectionHandler::DestructAllHandlers() {
749 std::set
<RTCPeerConnectionHandler
*> handlers(
750 g_peer_connection_handlers
.Get().begin(),
751 g_peer_connection_handlers
.Get().end());
752 for (auto handler
: handlers
) {
753 if (handler
->client_
)
754 handler
->client_
->releasePeerConnectionHandler();
759 void RTCPeerConnectionHandler::ConvertOfferOptionsToConstraints(
760 const blink::WebRTCOfferOptions
& options
,
761 RTCMediaConstraints
* output
) {
762 output
->AddMandatory(
763 webrtc::MediaConstraintsInterface::kOfferToReceiveAudio
,
764 options
.offerToReceiveAudio() > 0 ? "true" : "false",
767 output
->AddMandatory(
768 webrtc::MediaConstraintsInterface::kOfferToReceiveVideo
,
769 options
.offerToReceiveVideo() > 0 ? "true" : "false",
772 if (!options
.voiceActivityDetection()) {
773 output
->AddMandatory(
774 webrtc::MediaConstraintsInterface::kVoiceActivityDetection
,
779 if (options
.iceRestart()) {
780 output
->AddMandatory(
781 webrtc::MediaConstraintsInterface::kIceRestart
, "true", true);
785 void RTCPeerConnectionHandler::associateWithFrame(blink::WebFrame
* frame
) {
786 DCHECK(thread_checker_
.CalledOnValidThread());
791 bool RTCPeerConnectionHandler::initialize(
792 const blink::WebRTCConfiguration
& server_configuration
,
793 const blink::WebMediaConstraints
& options
) {
794 DCHECK(thread_checker_
.CalledOnValidThread());
797 peer_connection_tracker_
=
798 RenderThreadImpl::current()->peer_connection_tracker()->AsWeakPtr();
800 webrtc::PeerConnectionInterface::RTCConfiguration config
;
801 GetNativeRtcConfiguration(server_configuration
, &config
);
803 RTCMediaConstraints
constraints(options
);
805 peer_connection_observer_
= new Observer(weak_factory_
.GetWeakPtr());
806 native_peer_connection_
= dependency_factory_
->CreatePeerConnection(
807 config
, &constraints
, frame_
, peer_connection_observer_
.get());
809 if (!native_peer_connection_
.get()) {
810 LOG(ERROR
) << "Failed to initialize native PeerConnection.";
814 if (peer_connection_tracker_
) {
815 peer_connection_tracker_
->RegisterPeerConnection(
816 this, config
, constraints
, frame_
);
819 uma_observer_
= new rtc::RefCountedObject
<PeerConnectionUMAObserver
>();
820 native_peer_connection_
->RegisterUMAObserver(uma_observer_
.get());
824 bool RTCPeerConnectionHandler::InitializeForTest(
825 const blink::WebRTCConfiguration
& server_configuration
,
826 const blink::WebMediaConstraints
& options
,
827 const base::WeakPtr
<PeerConnectionTracker
>& peer_connection_tracker
) {
828 DCHECK(thread_checker_
.CalledOnValidThread());
829 webrtc::PeerConnectionInterface::RTCConfiguration config
;
830 GetNativeRtcConfiguration(server_configuration
, &config
);
832 peer_connection_observer_
= new Observer(weak_factory_
.GetWeakPtr());
833 RTCMediaConstraints
constraints(options
);
834 native_peer_connection_
= dependency_factory_
->CreatePeerConnection(
835 config
, &constraints
, NULL
, peer_connection_observer_
.get());
836 if (!native_peer_connection_
.get()) {
837 LOG(ERROR
) << "Failed to initialize native PeerConnection.";
840 peer_connection_tracker_
= peer_connection_tracker
;
844 void RTCPeerConnectionHandler::createOffer(
845 const blink::WebRTCSessionDescriptionRequest
& request
,
846 const blink::WebMediaConstraints
& options
) {
847 DCHECK(thread_checker_
.CalledOnValidThread());
848 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createOffer");
850 scoped_refptr
<CreateSessionDescriptionRequest
> description_request(
851 new rtc::RefCountedObject
<CreateSessionDescriptionRequest
>(
852 base::ThreadTaskRunnerHandle::Get(), request
,
853 weak_factory_
.GetWeakPtr(), peer_connection_tracker_
,
854 PeerConnectionTracker::ACTION_CREATE_OFFER
));
856 // TODO(tommi): Do this asynchronously via e.g. PostTaskAndReply.
857 RTCMediaConstraints
constraints(options
);
858 native_peer_connection_
->CreateOffer(description_request
.get(), &constraints
);
860 if (peer_connection_tracker_
)
861 peer_connection_tracker_
->TrackCreateOffer(this, constraints
);
864 void RTCPeerConnectionHandler::createOffer(
865 const blink::WebRTCSessionDescriptionRequest
& request
,
866 const blink::WebRTCOfferOptions
& options
) {
867 DCHECK(thread_checker_
.CalledOnValidThread());
868 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createOffer");
870 scoped_refptr
<CreateSessionDescriptionRequest
> description_request(
871 new rtc::RefCountedObject
<CreateSessionDescriptionRequest
>(
872 base::ThreadTaskRunnerHandle::Get(), request
,
873 weak_factory_
.GetWeakPtr(), peer_connection_tracker_
,
874 PeerConnectionTracker::ACTION_CREATE_OFFER
));
876 // TODO(tommi): Do this asynchronously via e.g. PostTaskAndReply.
877 RTCMediaConstraints constraints
;
878 ConvertOfferOptionsToConstraints(options
, &constraints
);
879 native_peer_connection_
->CreateOffer(description_request
.get(), &constraints
);
881 if (peer_connection_tracker_
)
882 peer_connection_tracker_
->TrackCreateOffer(this, constraints
);
885 void RTCPeerConnectionHandler::createAnswer(
886 const blink::WebRTCSessionDescriptionRequest
& request
,
887 const blink::WebMediaConstraints
& options
) {
888 DCHECK(thread_checker_
.CalledOnValidThread());
889 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createAnswer");
890 scoped_refptr
<CreateSessionDescriptionRequest
> description_request(
891 new rtc::RefCountedObject
<CreateSessionDescriptionRequest
>(
892 base::ThreadTaskRunnerHandle::Get(), request
,
893 weak_factory_
.GetWeakPtr(), peer_connection_tracker_
,
894 PeerConnectionTracker::ACTION_CREATE_ANSWER
));
895 // TODO(tommi): Do this asynchronously via e.g. PostTaskAndReply.
896 RTCMediaConstraints
constraints(options
);
897 native_peer_connection_
->CreateAnswer(description_request
.get(),
900 if (peer_connection_tracker_
)
901 peer_connection_tracker_
->TrackCreateAnswer(this, constraints
);
904 void RTCPeerConnectionHandler::setLocalDescription(
905 const blink::WebRTCVoidRequest
& request
,
906 const blink::WebRTCSessionDescription
& description
) {
907 DCHECK(thread_checker_
.CalledOnValidThread());
908 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::setLocalDescription");
910 std::string sdp
= base::UTF16ToUTF8(base::StringPiece16(description
.sdp()));
912 base::UTF16ToUTF8(base::StringPiece16(description
.type()));
914 webrtc::SdpParseError error
;
915 // Since CreateNativeSessionDescription uses the dependency factory, we need
916 // to make this call on the current thread to be safe.
917 webrtc::SessionDescriptionInterface
* native_desc
=
918 CreateNativeSessionDescription(sdp
, type
, &error
);
920 std::string reason_str
= "Failed to parse SessionDescription. ";
921 reason_str
.append(error
.line
);
922 reason_str
.append(" ");
923 reason_str
.append(error
.description
);
924 LOG(ERROR
) << reason_str
;
925 request
.requestFailed(blink::WebString::fromUTF8(reason_str
));
929 if (peer_connection_tracker_
) {
930 peer_connection_tracker_
->TrackSetSessionDescription(
931 this, sdp
, type
, PeerConnectionTracker::SOURCE_LOCAL
);
934 scoped_refptr
<SetSessionDescriptionRequest
> set_request(
935 new rtc::RefCountedObject
<SetSessionDescriptionRequest
>(
936 base::ThreadTaskRunnerHandle::Get(), request
,
937 weak_factory_
.GetWeakPtr(), peer_connection_tracker_
,
938 PeerConnectionTracker::ACTION_SET_LOCAL_DESCRIPTION
));
940 signaling_thread()->PostTask(FROM_HERE
,
941 base::Bind(&RunClosureWithTrace
,
942 base::Bind(&webrtc::PeerConnectionInterface::SetLocalDescription
,
943 native_peer_connection_
, set_request
,
944 base::Unretained(native_desc
)),
945 "SetLocalDescription"));
948 void RTCPeerConnectionHandler::setRemoteDescription(
949 const blink::WebRTCVoidRequest
& request
,
950 const blink::WebRTCSessionDescription
& description
) {
951 DCHECK(thread_checker_
.CalledOnValidThread());
952 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::setRemoteDescription");
953 std::string sdp
= base::UTF16ToUTF8(base::StringPiece16(description
.sdp()));
955 base::UTF16ToUTF8(base::StringPiece16(description
.type()));
957 webrtc::SdpParseError error
;
958 // Since CreateNativeSessionDescription uses the dependency factory, we need
959 // to make this call on the current thread to be safe.
960 webrtc::SessionDescriptionInterface
* native_desc
=
961 CreateNativeSessionDescription(sdp
, type
, &error
);
963 std::string reason_str
= "Failed to parse SessionDescription. ";
964 reason_str
.append(error
.line
);
965 reason_str
.append(" ");
966 reason_str
.append(error
.description
);
967 LOG(ERROR
) << reason_str
;
968 request
.requestFailed(blink::WebString::fromUTF8(reason_str
));
972 if (peer_connection_tracker_
) {
973 peer_connection_tracker_
->TrackSetSessionDescription(
974 this, sdp
, type
, PeerConnectionTracker::SOURCE_REMOTE
);
977 scoped_refptr
<SetSessionDescriptionRequest
> set_request(
978 new rtc::RefCountedObject
<SetSessionDescriptionRequest
>(
979 base::ThreadTaskRunnerHandle::Get(), request
,
980 weak_factory_
.GetWeakPtr(), peer_connection_tracker_
,
981 PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION
));
982 signaling_thread()->PostTask(FROM_HERE
,
983 base::Bind(&RunClosureWithTrace
,
984 base::Bind(&webrtc::PeerConnectionInterface::SetRemoteDescription
,
985 native_peer_connection_
, set_request
,
986 base::Unretained(native_desc
)),
987 "SetRemoteDescription"));
990 blink::WebRTCSessionDescription
991 RTCPeerConnectionHandler::localDescription() {
992 DCHECK(thread_checker_
.CalledOnValidThread());
993 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::localDescription");
995 // Since local_description returns a pointer to a non-reference-counted object
996 // that lives on the signaling thread, we cannot fetch a pointer to it and use
997 // it directly here. Instead, we access the object completely on the signaling
999 std::string sdp
, type
;
1000 base::Callback
<const webrtc::SessionDescriptionInterface
*()> description_cb
=
1001 base::Bind(&webrtc::PeerConnectionInterface::local_description
,
1002 native_peer_connection_
);
1003 RunSynchronousClosureOnSignalingThread(
1004 base::Bind(&GetSdpAndTypeFromSessionDescription
, description_cb
,
1005 base::Unretained(&sdp
), base::Unretained(&type
)),
1006 "localDescription");
1008 return CreateWebKitSessionDescription(sdp
, type
);
1011 blink::WebRTCSessionDescription
1012 RTCPeerConnectionHandler::remoteDescription() {
1013 DCHECK(thread_checker_
.CalledOnValidThread());
1014 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::remoteDescription");
1015 // Since local_description returns a pointer to a non-reference-counted object
1016 // that lives on the signaling thread, we cannot fetch a pointer to it and use
1017 // it directly here. Instead, we access the object completely on the signaling
1019 std::string sdp
, type
;
1020 base::Callback
<const webrtc::SessionDescriptionInterface
*()> description_cb
=
1021 base::Bind(&webrtc::PeerConnectionInterface::remote_description
,
1022 native_peer_connection_
);
1023 RunSynchronousClosureOnSignalingThread(
1024 base::Bind(&GetSdpAndTypeFromSessionDescription
, description_cb
,
1025 base::Unretained(&sdp
), base::Unretained(&type
)),
1026 "remoteDescription");
1028 return CreateWebKitSessionDescription(sdp
, type
);
1031 bool RTCPeerConnectionHandler::updateICE(
1032 const blink::WebRTCConfiguration
& server_configuration
,
1033 const blink::WebMediaConstraints
& options
) {
1034 DCHECK(thread_checker_
.CalledOnValidThread());
1035 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::updateICE");
1036 webrtc::PeerConnectionInterface::RTCConfiguration config
;
1037 GetNativeRtcConfiguration(server_configuration
, &config
);
1038 RTCMediaConstraints
constraints(options
);
1040 if (peer_connection_tracker_
)
1041 peer_connection_tracker_
->TrackUpdateIce(this, config
, constraints
);
1043 return native_peer_connection_
->UpdateIce(config
.servers
, &constraints
);
1046 bool RTCPeerConnectionHandler::addICECandidate(
1047 const blink::WebRTCVoidRequest
& request
,
1048 const blink::WebRTCICECandidate
& candidate
) {
1049 DCHECK(thread_checker_
.CalledOnValidThread());
1050 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::addICECandidate");
1051 // Libjingle currently does not accept callbacks for addICECandidate.
1052 // For that reason we are going to call callbacks from here.
1054 // TODO(tommi): Instead of calling addICECandidate here, we can do a
1055 // PostTaskAndReply kind of a thing.
1056 bool result
= addICECandidate(candidate
);
1057 base::ThreadTaskRunnerHandle::Get()->PostTask(
1058 FROM_HERE
, base::Bind(&RTCPeerConnectionHandler::OnaddICECandidateResult
,
1059 weak_factory_
.GetWeakPtr(), request
, result
));
1060 // On failure callback will be triggered.
1064 bool RTCPeerConnectionHandler::addICECandidate(
1065 const blink::WebRTCICECandidate
& candidate
) {
1066 DCHECK(thread_checker_
.CalledOnValidThread());
1067 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::addICECandidate");
1068 scoped_ptr
<webrtc::IceCandidateInterface
> native_candidate(
1069 dependency_factory_
->CreateIceCandidate(
1070 base::UTF16ToUTF8(base::StringPiece16(candidate
.sdpMid())),
1071 candidate
.sdpMLineIndex(),
1072 base::UTF16ToUTF8(base::StringPiece16(candidate
.candidate()))));
1073 bool return_value
= false;
1075 if (native_candidate
) {
1077 native_peer_connection_
->AddIceCandidate(native_candidate
.get());
1078 LOG_IF(ERROR
, !return_value
) << "Error processing ICE candidate.";
1080 LOG(ERROR
) << "Could not create native ICE candidate.";
1083 if (peer_connection_tracker_
) {
1084 peer_connection_tracker_
->TrackAddIceCandidate(
1085 this, candidate
, PeerConnectionTracker::SOURCE_REMOTE
, return_value
);
1087 return return_value
;
1090 void RTCPeerConnectionHandler::OnaddICECandidateResult(
1091 const blink::WebRTCVoidRequest
& webkit_request
, bool result
) {
1092 DCHECK(thread_checker_
.CalledOnValidThread());
1093 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnaddICECandidateResult");
1095 // We don't have the actual error code from the libjingle, so for now
1096 // using a generic error string.
1097 return webkit_request
.requestFailed(
1098 base::UTF8ToUTF16("Error processing ICE candidate"));
1101 return webkit_request
.requestSucceeded();
1104 bool RTCPeerConnectionHandler::addStream(
1105 const blink::WebMediaStream
& stream
,
1106 const blink::WebMediaConstraints
& options
) {
1107 DCHECK(thread_checker_
.CalledOnValidThread());
1108 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::addStream");
1109 for (ScopedVector
<WebRtcMediaStreamAdapter
>::iterator adapter_it
=
1110 local_streams_
.begin(); adapter_it
!= local_streams_
.end();
1112 if ((*adapter_it
)->IsEqual(stream
)) {
1113 DVLOG(1) << "RTCPeerConnectionHandler::addStream called with the same "
1114 << "stream twice. id=" << stream
.id().utf8();
1119 if (peer_connection_tracker_
) {
1120 peer_connection_tracker_
->TrackAddStream(
1121 this, stream
, PeerConnectionTracker::SOURCE_LOCAL
);
1124 PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
1126 WebRtcMediaStreamAdapter
* adapter
=
1127 new WebRtcMediaStreamAdapter(stream
, dependency_factory_
);
1128 local_streams_
.push_back(adapter
);
1130 webrtc::MediaStreamInterface
* webrtc_stream
= adapter
->webrtc_media_stream();
1131 track_metrics_
.AddStream(MediaStreamTrackMetrics::SENT_STREAM
,
1134 RTCMediaConstraints
constraints(options
);
1135 if (!constraints
.GetMandatory().empty() ||
1136 !constraints
.GetOptional().empty()) {
1137 // TODO(perkj): |mediaConstraints| is the name of the optional constraints
1138 // argument in RTCPeerConnection.idl. It has been removed from the spec and
1139 // should be removed from blink as well.
1141 << "mediaConstraints is not a supported argument to addStream.";
1144 return native_peer_connection_
->AddStream(webrtc_stream
);
1147 void RTCPeerConnectionHandler::removeStream(
1148 const blink::WebMediaStream
& stream
) {
1149 DCHECK(thread_checker_
.CalledOnValidThread());
1150 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::removeStream");
1151 // Find the webrtc stream.
1152 scoped_refptr
<webrtc::MediaStreamInterface
> webrtc_stream
;
1153 for (ScopedVector
<WebRtcMediaStreamAdapter
>::iterator adapter_it
=
1154 local_streams_
.begin(); adapter_it
!= local_streams_
.end();
1156 if ((*adapter_it
)->IsEqual(stream
)) {
1157 webrtc_stream
= (*adapter_it
)->webrtc_media_stream();
1158 local_streams_
.erase(adapter_it
);
1162 DCHECK(webrtc_stream
.get());
1163 // TODO(tommi): Make this async (PostTaskAndReply).
1164 native_peer_connection_
->RemoveStream(webrtc_stream
.get());
1166 if (peer_connection_tracker_
) {
1167 peer_connection_tracker_
->TrackRemoveStream(
1168 this, stream
, PeerConnectionTracker::SOURCE_LOCAL
);
1170 PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
1171 track_metrics_
.RemoveStream(MediaStreamTrackMetrics::SENT_STREAM
,
1172 webrtc_stream
.get());
1175 void RTCPeerConnectionHandler::getStats(
1176 const blink::WebRTCStatsRequest
& request
) {
1177 DCHECK(thread_checker_
.CalledOnValidThread());
1178 scoped_refptr
<LocalRTCStatsRequest
> inner_request(
1179 new rtc::RefCountedObject
<LocalRTCStatsRequest
>(request
));
1180 getStats(inner_request
);
1183 void RTCPeerConnectionHandler::getStats(
1184 const scoped_refptr
<LocalRTCStatsRequest
>& request
) {
1185 DCHECK(thread_checker_
.CalledOnValidThread());
1186 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::getStats");
1189 rtc::scoped_refptr
<webrtc::StatsObserver
> observer(
1190 new rtc::RefCountedObject
<StatsResponse
>(request
));
1192 std::string track_id
;
1193 blink::WebMediaStreamSource::Type track_type
=
1194 blink::WebMediaStreamSource::TypeAudio
;
1195 if (request
->hasSelector()) {
1196 track_type
= request
->component().source().type();
1197 track_id
= request
->component().id().utf8();
1200 GetStats(observer
, webrtc::PeerConnectionInterface::kStatsOutputLevelStandard
,
1201 track_id
, track_type
);
1204 // TODO(tommi): It's weird to have three {g|G}etStats methods. Clean this up.
1205 void RTCPeerConnectionHandler::GetStats(
1206 webrtc::StatsObserver
* observer
,
1207 webrtc::PeerConnectionInterface::StatsOutputLevel level
,
1208 const std::string
& track_id
,
1209 blink::WebMediaStreamSource::Type track_type
) {
1210 DCHECK(thread_checker_
.CalledOnValidThread());
1211 signaling_thread()->PostTask(FROM_HERE
,
1212 base::Bind(&GetStatsOnSignalingThread
, native_peer_connection_
, level
,
1213 make_scoped_refptr(observer
), track_id
, track_type
));
1216 void RTCPeerConnectionHandler::CloseClientPeerConnection() {
1217 DCHECK(thread_checker_
.CalledOnValidThread());
1219 client_
->closePeerConnection();
1222 blink::WebRTCDataChannelHandler
* RTCPeerConnectionHandler::createDataChannel(
1223 const blink::WebString
& label
, const blink::WebRTCDataChannelInit
& init
) {
1224 DCHECK(thread_checker_
.CalledOnValidThread());
1225 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDataChannel");
1226 DVLOG(1) << "createDataChannel label "
1227 << base::UTF16ToUTF8(base::StringPiece16(label
));
1229 webrtc::DataChannelInit config
;
1230 // TODO(jiayl): remove the deprecated reliable field once Libjingle is updated
1232 config
.reliable
= false;
1233 config
.id
= init
.id
;
1234 config
.ordered
= init
.ordered
;
1235 config
.negotiated
= init
.negotiated
;
1236 config
.maxRetransmits
= init
.maxRetransmits
;
1237 config
.maxRetransmitTime
= init
.maxRetransmitTime
;
1238 config
.protocol
= base::UTF16ToUTF8(base::StringPiece16(init
.protocol
));
1240 rtc::scoped_refptr
<webrtc::DataChannelInterface
> webrtc_channel(
1241 native_peer_connection_
->CreateDataChannel(
1242 base::UTF16ToUTF8(base::StringPiece16(label
)), &config
));
1243 if (!webrtc_channel
) {
1244 DLOG(ERROR
) << "Could not create native data channel.";
1247 if (peer_connection_tracker_
) {
1248 peer_connection_tracker_
->TrackCreateDataChannel(
1249 this, webrtc_channel
.get(), PeerConnectionTracker::SOURCE_LOCAL
);
1252 ++num_data_channels_created_
;
1254 return new RtcDataChannelHandler(base::ThreadTaskRunnerHandle::Get(),
1258 blink::WebRTCDTMFSenderHandler
* RTCPeerConnectionHandler::createDTMFSender(
1259 const blink::WebMediaStreamTrack
& track
) {
1260 DCHECK(thread_checker_
.CalledOnValidThread());
1261 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender");
1262 DVLOG(1) << "createDTMFSender.";
1264 MediaStreamTrack
* native_track
= MediaStreamTrack::GetTrack(track
);
1265 if (!native_track
|| !native_track
->is_local_track() ||
1266 track
.source().type() != blink::WebMediaStreamSource::TypeAudio
) {
1267 DLOG(ERROR
) << "The DTMF sender requires a local audio track.";
1271 scoped_refptr
<webrtc::AudioTrackInterface
> audio_track
=
1272 native_track
->GetAudioAdapter();
1273 rtc::scoped_refptr
<webrtc::DtmfSenderInterface
> sender(
1274 native_peer_connection_
->CreateDtmfSender(audio_track
.get()));
1276 DLOG(ERROR
) << "Could not create native DTMF sender.";
1279 if (peer_connection_tracker_
)
1280 peer_connection_tracker_
->TrackCreateDTMFSender(this, track
);
1282 return new RtcDtmfSenderHandler(sender
);
1285 void RTCPeerConnectionHandler::stop() {
1286 DCHECK(thread_checker_
.CalledOnValidThread());
1287 DVLOG(1) << "RTCPeerConnectionHandler::stop";
1289 if (!client_
|| !native_peer_connection_
.get())
1290 return; // Already stopped.
1292 if (peer_connection_tracker_
)
1293 peer_connection_tracker_
->TrackStop(this);
1295 native_peer_connection_
->Close();
1297 // The client_ pointer is not considered valid after this point and no further
1298 // callbacks must be made.
1302 void RTCPeerConnectionHandler::OnSignalingChange(
1303 webrtc::PeerConnectionInterface::SignalingState new_state
) {
1304 DCHECK(thread_checker_
.CalledOnValidThread());
1305 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnSignalingChange");
1307 blink::WebRTCPeerConnectionHandlerClient::SignalingState state
=
1308 GetWebKitSignalingState(new_state
);
1309 if (peer_connection_tracker_
)
1310 peer_connection_tracker_
->TrackSignalingStateChange(this, state
);
1312 client_
->didChangeSignalingState(state
);
1315 // Called any time the IceConnectionState changes
1316 void RTCPeerConnectionHandler::OnIceConnectionChange(
1317 webrtc::PeerConnectionInterface::IceConnectionState new_state
) {
1318 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnIceConnectionChange");
1319 DCHECK(thread_checker_
.CalledOnValidThread());
1320 if (new_state
== webrtc::PeerConnectionInterface::kIceConnectionChecking
) {
1321 ice_connection_checking_start_
= base::TimeTicks::Now();
1322 } else if (new_state
==
1323 webrtc::PeerConnectionInterface::kIceConnectionConnected
) {
1324 // If the state becomes connected, send the time needed for PC to become
1325 // connected from checking to UMA. UMA data will help to know how much
1326 // time needed for PC to connect with remote peer.
1327 if (ice_connection_checking_start_
.is_null()) {
1328 // From UMA, we have observed a large number of calls falling into the
1329 // overflow buckets. One possibility is that the Checking is not signaled
1330 // before Connected. This is to guard against that situation to make the
1331 // metric more robust.
1332 UMA_HISTOGRAM_MEDIUM_TIMES("WebRTC.PeerConnection.TimeToConnect",
1335 UMA_HISTOGRAM_MEDIUM_TIMES(
1336 "WebRTC.PeerConnection.TimeToConnect",
1337 base::TimeTicks::Now() - ice_connection_checking_start_
);
1341 track_metrics_
.IceConnectionChange(new_state
);
1342 blink::WebRTCPeerConnectionHandlerClient::ICEConnectionState state
=
1343 GetWebKitIceConnectionState(new_state
);
1344 if (peer_connection_tracker_
)
1345 peer_connection_tracker_
->TrackIceConnectionStateChange(this, state
);
1347 client_
->didChangeICEConnectionState(state
);
1350 // Called any time the IceGatheringState changes
1351 void RTCPeerConnectionHandler::OnIceGatheringChange(
1352 webrtc::PeerConnectionInterface::IceGatheringState new_state
) {
1353 DCHECK(thread_checker_
.CalledOnValidThread());
1354 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnIceGatheringChange");
1356 if (new_state
== webrtc::PeerConnectionInterface::kIceGatheringComplete
) {
1357 // If ICE gathering is completed, generate a NULL ICE candidate,
1358 // to signal end of candidates.
1360 blink::WebRTCICECandidate null_candidate
;
1361 client_
->didGenerateICECandidate(null_candidate
);
1364 UMA_HISTOGRAM_COUNTS_100("WebRTC.PeerConnection.IPv4LocalCandidates",
1365 num_local_candidates_ipv4_
);
1367 UMA_HISTOGRAM_COUNTS_100("WebRTC.PeerConnection.IPv6LocalCandidates",
1368 num_local_candidates_ipv6_
);
1369 } else if (new_state
==
1370 webrtc::PeerConnectionInterface::kIceGatheringGathering
) {
1371 // ICE restarts will change gathering state back to "gathering",
1372 // reset the counter.
1373 num_local_candidates_ipv6_
= 0;
1374 num_local_candidates_ipv4_
= 0;
1377 blink::WebRTCPeerConnectionHandlerClient::ICEGatheringState state
=
1378 GetWebKitIceGatheringState(new_state
);
1379 if (peer_connection_tracker_
)
1380 peer_connection_tracker_
->TrackIceGatheringStateChange(this, state
);
1382 client_
->didChangeICEGatheringState(state
);
1385 void RTCPeerConnectionHandler::OnRenegotiationNeeded() {
1386 DCHECK(thread_checker_
.CalledOnValidThread());
1387 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnRenegotiationNeeded");
1388 if (peer_connection_tracker_
)
1389 peer_connection_tracker_
->TrackOnRenegotiationNeeded(this);
1391 client_
->negotiationNeeded();
1394 void RTCPeerConnectionHandler::OnAddStream(
1395 scoped_ptr
<RemoteMediaStreamImpl
> stream
) {
1396 DCHECK(thread_checker_
.CalledOnValidThread());
1397 DCHECK(remote_streams_
.find(stream
->webrtc_stream().get()) ==
1398 remote_streams_
.end());
1399 DCHECK(stream
->webkit_stream().extraData()) << "Initialization not done";
1400 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnAddStreamImpl");
1402 // Ownership is with remote_streams_ now.
1403 RemoteMediaStreamImpl
* s
= stream
.release();
1404 remote_streams_
.insert(
1405 std::pair
<webrtc::MediaStreamInterface
*, RemoteMediaStreamImpl
*> (
1406 s
->webrtc_stream().get(), s
));
1408 if (peer_connection_tracker_
) {
1409 peer_connection_tracker_
->TrackAddStream(
1410 this, s
->webkit_stream(), PeerConnectionTracker::SOURCE_REMOTE
);
1413 PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
1415 track_metrics_
.AddStream(MediaStreamTrackMetrics::RECEIVED_STREAM
,
1416 s
->webrtc_stream().get());
1418 client_
->didAddRemoteStream(s
->webkit_stream());
1421 void RTCPeerConnectionHandler::OnRemoveStream(
1422 const scoped_refptr
<webrtc::MediaStreamInterface
>& stream
) {
1423 DCHECK(thread_checker_
.CalledOnValidThread());
1424 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnRemoveStreamImpl");
1425 RemoteStreamMap::iterator it
= remote_streams_
.find(stream
.get());
1426 if (it
== remote_streams_
.end()) {
1427 NOTREACHED() << "Stream not found";
1431 track_metrics_
.RemoveStream(MediaStreamTrackMetrics::RECEIVED_STREAM
,
1433 PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
1435 scoped_ptr
<RemoteMediaStreamImpl
> remote_stream(it
->second
);
1436 const blink::WebMediaStream
& webkit_stream
= remote_stream
->webkit_stream();
1437 DCHECK(!webkit_stream
.isNull());
1438 remote_streams_
.erase(it
);
1440 if (peer_connection_tracker_
) {
1441 peer_connection_tracker_
->TrackRemoveStream(
1442 this, webkit_stream
, PeerConnectionTracker::SOURCE_REMOTE
);
1446 client_
->didRemoveRemoteStream(webkit_stream
);
1449 void RTCPeerConnectionHandler::OnDataChannel(
1450 scoped_ptr
<RtcDataChannelHandler
> handler
) {
1451 DCHECK(thread_checker_
.CalledOnValidThread());
1452 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnDataChannelImpl");
1454 if (peer_connection_tracker_
) {
1455 peer_connection_tracker_
->TrackCreateDataChannel(
1456 this, handler
->channel().get(), PeerConnectionTracker::SOURCE_REMOTE
);
1460 client_
->didAddRemoteDataChannel(handler
.release());
1463 void RTCPeerConnectionHandler::OnIceCandidate(
1464 const std::string
& sdp
, const std::string
& sdp_mid
, int sdp_mline_index
,
1465 int component
, int address_family
) {
1466 DCHECK(thread_checker_
.CalledOnValidThread());
1467 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnIceCandidateImpl");
1468 blink::WebRTCICECandidate web_candidate
;
1469 web_candidate
.initialize(base::UTF8ToUTF16(sdp
),
1470 base::UTF8ToUTF16(sdp_mid
),
1472 if (peer_connection_tracker_
) {
1473 peer_connection_tracker_
->TrackAddIceCandidate(
1474 this, web_candidate
, PeerConnectionTracker::SOURCE_LOCAL
, true);
1477 // Only the first m line's first component is tracked to avoid
1478 // miscounting when doing BUNDLE or rtcp mux.
1479 if (sdp_mline_index
== 0 && component
== 1) {
1480 if (address_family
== AF_INET
) {
1481 ++num_local_candidates_ipv4_
;
1482 } else if (address_family
== AF_INET6
) {
1483 ++num_local_candidates_ipv6_
;
1489 client_
->didGenerateICECandidate(web_candidate
);
1492 webrtc::SessionDescriptionInterface
*
1493 RTCPeerConnectionHandler::CreateNativeSessionDescription(
1494 const std::string
& sdp
, const std::string
& type
,
1495 webrtc::SdpParseError
* error
) {
1496 webrtc::SessionDescriptionInterface
* native_desc
=
1497 dependency_factory_
->CreateSessionDescription(type
, sdp
, error
);
1499 LOG_IF(ERROR
, !native_desc
) << "Failed to create native session description."
1500 << " Type: " << type
<< " SDP: " << sdp
;
1505 scoped_refptr
<base::SingleThreadTaskRunner
>
1506 RTCPeerConnectionHandler::signaling_thread() const {
1507 DCHECK(thread_checker_
.CalledOnValidThread());
1508 return dependency_factory_
->GetWebRtcSignalingThread();
1511 void RTCPeerConnectionHandler::RunSynchronousClosureOnSignalingThread(
1512 const base::Closure
& closure
,
1513 const char* trace_event_name
) {
1514 DCHECK(thread_checker_
.CalledOnValidThread());
1515 scoped_refptr
<base::SingleThreadTaskRunner
> thread(signaling_thread());
1516 if (!thread
.get() || thread
->BelongsToCurrentThread()) {
1517 TRACE_EVENT0("webrtc", trace_event_name
);
1520 base::WaitableEvent
event(false, false);
1521 thread
->PostTask(FROM_HERE
,
1522 base::Bind(&RunSynchronousClosure
, closure
,
1523 base::Unretained(trace_event_name
),
1524 base::Unretained(&event
)));
1529 } // namespace content