Revert "Use AsyncInvoker in PeerConnection instead of MessageHandler"
This reverts commit bb19276a325a5f9fce4afa245aa14ec2a4b1a41d.
Reason for revert: breaks downstream project
Original change's description:
> Use AsyncInvoker in PeerConnection instead of MessageHandler
>
> Bug: webrtc:9702
> Change-Id: I89d66d1165a096601aed37b8febad60620073899
> Reviewed-on: https://webrtc-review.googlesource.com/97180
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24515}
TBR=steveanton@webrtc.org,shampson@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9702
Change-Id: Ibfe507cd1593f7000e11f9a17313a016307381cb
Reviewed-on: https://webrtc-review.googlesource.com/98302
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24591}diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 4b6eb17..6c543f9 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -100,8 +100,43 @@
// The length of RTCP CNAMEs.
static const int kRtcpCnameLength = 16;
+enum {
+ MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
+ MSG_SET_SESSIONDESCRIPTION_FAILED,
+ MSG_CREATE_SESSIONDESCRIPTION_FAILED,
+ MSG_GETSTATS,
+ MSG_FREE_DATACHANNELS,
+ MSG_REPORT_USAGE_PATTERN,
+};
+
static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
+struct SetSessionDescriptionMsg : public rtc::MessageData {
+ explicit SetSessionDescriptionMsg(
+ webrtc::SetSessionDescriptionObserver* observer)
+ : observer(observer) {}
+
+ rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
+ RTCError error;
+};
+
+struct CreateSessionDescriptionMsg : public rtc::MessageData {
+ explicit CreateSessionDescriptionMsg(
+ webrtc::CreateSessionDescriptionObserver* observer)
+ : observer(observer) {}
+
+ rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
+ RTCError error;
+};
+
+struct GetStatsMsg : public rtc::MessageData {
+ GetStatsMsg(webrtc::StatsObserver* observer,
+ webrtc::MediaStreamTrackInterface* track)
+ : observer(observer), track(track) {}
+ rtc::scoped_refptr<webrtc::StatsObserver> observer;
+ rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
+};
+
// Check if we can send |new_stream| on a PeerConnection.
bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
webrtc::MediaStreamInterface* new_stream) {
@@ -992,9 +1027,8 @@
}
int delay_ms =
return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
- async_invoker_.AsyncInvokeDelayed<void>(RTC_FROM_HERE, signaling_thread(),
- [this] { ReportUsagePattern(); },
- delay_ms);
+ signaling_thread()->PostDelayed(RTC_FROM_HERE, delay_ms, this,
+ MSG_REPORT_USAGE_PATTERN, nullptr);
return true;
}
@@ -1547,16 +1581,8 @@
<< track->id();
return false;
}
- // Need to capture |observer| and |track| in scoped_refptrs to ensure they
- // live long enough.
- rtc::scoped_refptr<StatsObserver> observer_refptr(observer);
- rtc::scoped_refptr<MediaStreamTrackInterface> track_refptr(track);
- async_invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(),
- [this, observer_refptr, track_refptr] {
- StatsReports reports;
- stats_->GetStats(track_refptr, &reports);
- observer_refptr->OnComplete(reports);
- });
+ signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
+ new GetStatsMsg(observer, track));
return true;
}
@@ -1885,9 +1911,9 @@
PostSetSessionDescriptionSuccess(observer);
- // MaybeStartGathering needs to be called after posting OnSuccess to the
- // SetSessionDescriptionObserver so that we don't signal any candidates before
- // signaling that SetLocalDescription completed.
+ // MaybeStartGathering needs to be called after posting
+ // MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
+ // before signaling that SetLocalDescription completed.
transport_controller_->MaybeStartGathering();
if (local_description()->GetType() == SdpType::kAnswer) {
@@ -3210,6 +3236,51 @@
observer_ = nullptr;
}
+void PeerConnection::OnMessage(rtc::Message* msg) {
+ switch (msg->message_id) {
+ case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
+ SetSessionDescriptionMsg* param =
+ static_cast<SetSessionDescriptionMsg*>(msg->pdata);
+ param->observer->OnSuccess();
+ delete param;
+ break;
+ }
+ case MSG_SET_SESSIONDESCRIPTION_FAILED: {
+ SetSessionDescriptionMsg* param =
+ static_cast<SetSessionDescriptionMsg*>(msg->pdata);
+ param->observer->OnFailure(std::move(param->error));
+ delete param;
+ break;
+ }
+ case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
+ CreateSessionDescriptionMsg* param =
+ static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
+ param->observer->OnFailure(std::move(param->error));
+ delete param;
+ break;
+ }
+ case MSG_GETSTATS: {
+ GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
+ StatsReports reports;
+ stats_->GetStats(param->track, &reports);
+ param->observer->OnComplete(reports);
+ delete param;
+ break;
+ }
+ case MSG_FREE_DATACHANNELS: {
+ sctp_data_channels_to_free_.clear();
+ break;
+ }
+ case MSG_REPORT_USAGE_PATTERN: {
+ ReportUsagePattern();
+ break;
+ }
+ default:
+ RTC_NOTREACHED() << "Not implemented";
+ break;
+ }
+}
+
cricket::VoiceMediaChannel* PeerConnection::voice_media_channel() const {
RTC_DCHECK(!IsUnifiedPlan());
auto* voice_channel = static_cast<cricket::VoiceChannel*>(
@@ -3482,37 +3553,29 @@
void PeerConnection::PostSetSessionDescriptionSuccess(
SetSessionDescriptionObserver* observer) {
- async_invoker_.AsyncInvoke<void>(
- RTC_FROM_HERE, signaling_thread(),
- rtc::Bind(&SetSessionDescriptionObserver::OnSuccess, observer));
+ SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
+ signaling_thread()->Post(RTC_FROM_HERE, this,
+ MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
}
void PeerConnection::PostSetSessionDescriptionFailure(
SetSessionDescriptionObserver* observer,
- RTCError error) {
+ RTCError&& error) {
RTC_DCHECK(!error.ok());
- // TODO(steveanton): In C++14 this can be done with a lambda.
- struct Functor {
- void operator()() { observer->OnFailure(std::move(error)); }
- rtc::scoped_refptr<SetSessionDescriptionObserver> observer;
- RTCError error;
- };
- async_invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(),
- Functor{observer, std::move(error)});
+ SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
+ msg->error = std::move(error);
+ signaling_thread()->Post(RTC_FROM_HERE, this,
+ MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
}
void PeerConnection::PostCreateSessionDescriptionFailure(
CreateSessionDescriptionObserver* observer,
RTCError error) {
RTC_DCHECK(!error.ok());
- // TODO(steveanton): In C++14 this can be done with a lambda.
- struct Functor {
- void operator()() { observer->OnFailure(std::move(error)); }
- rtc::scoped_refptr<CreateSessionDescriptionObserver> observer;
- RTCError error;
- };
- async_invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(),
- Functor{observer, std::move(error)});
+ CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
+ msg->error = std::move(error);
+ signaling_thread()->Post(RTC_FROM_HERE, this,
+ MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
}
void PeerConnection::GetOptionsForOffer(
@@ -4404,9 +4467,8 @@
// we can't free it directly here; we need to free it asynchronously.
sctp_data_channels_to_free_.push_back(*it);
sctp_data_channels_.erase(it);
- async_invoker_.AsyncInvoke<void>(
- RTC_FROM_HERE, signaling_thread(),
- [this] { sctp_data_channels_to_free_.clear(); });
+ signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
+ nullptr);
return;
}
}
@@ -6170,8 +6232,8 @@
}
void PeerConnection::RequestUsagePatternReportForTesting() {
- async_invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(),
- [this] { ReportUsagePattern(); });
+ signaling_thread()->Post(RTC_FROM_HERE, this, MSG_REPORT_USAGE_PATTERN,
+ nullptr);
}
} // namespace webrtc
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index bd9ef24..2942709 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -52,6 +52,7 @@
class PeerConnection : public PeerConnectionInternal,
public DataChannelProviderInterface,
public JsepTransportController::Observer,
+ public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
enum class UsageEvent : int {
@@ -287,6 +288,9 @@
uint32_t first_ssrc;
};
+ // Implements MessageHandler.
+ void OnMessage(rtc::Message* msg) override;
+
// Plan B helpers for getting the voice/video media channels for the single
// audio/video transceiver, if it exists.
cricket::VoiceMediaChannel* voice_media_channel() const;
@@ -391,7 +395,7 @@
void PostSetSessionDescriptionSuccess(
SetSessionDescriptionObserver* observer);
void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
- RTCError error);
+ RTCError&& error);
void PostCreateSessionDescriptionFailure(
CreateSessionDescriptionObserver* observer,
RTCError error);
@@ -1029,7 +1033,6 @@
int usage_event_accumulator_ = 0;
bool return_histogram_very_quickly_ = false;
- rtc::AsyncInvoker async_invoker_;
};
} // namespace webrtc