webrtc_m130/pc/sctp_transport.cc
Tomas Gunnarsson 92eebefd47 Reland "Fix unsynchronized access to mid_to_transport_ in JsepTransportController"
This reverts commit 6b143c1c0686519bc9d73223c1350cee286c8d78.

Reason for revert:
  Relanding with updated expectations for SctpTransport::Information
  based on TransceiverStateSurfacer in Chromium.


Original change's description:
> Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController"
>
> This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4.
>
> Reason for revert: Breaks WebRTC Chromium FYI Bots
>
> First failure:
> https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
>
> Failed tests:
> WebRtcDataBrowserTest.CallWithSctpDataAndMedia
> WebRtcDataBrowserTest.CallWithSctpDataOnly
>
> Original change's description:
> > Fix unsynchronized access to mid_to_transport_ in JsepTransportController
> >
> > * Added several thread checks to JTC to help with programmer errors.
> > * Avoid a few Invokes() to the network thread here and there such
> >   as for fetching sctp transport name for getStats(). The transport
> >   name is now cached when it changes on the network thread.
> > * JsepTransportController instances now get deleted on the network
> >   thread rather than on the signaling thread + issuing an Invoke()
> >   in the dtor.
> > * Moved some thread hops from JTC over to PC which is where the problem
> >   exists and also (imho) makes it easier to see where hops happen in
> >   the PC code.
> > * The sctp transport is now started asynchronously when we push down the
> >   media description.
> > * PeerConnection proxy calls GetSctpTransport directly on the network
> >   thread instead of to the signaling thread + blocking on the network
> >   thread.
> > * The above changes simplified things for webrtc::SctpTransport which
> >   allowed for removing locking from that class and delete some code.
> >
> > Bug: webrtc:9987, webrtc:12445
> > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33191}
>
> TBR=tommi@webrtc.org,hta@webrtc.org
>
> Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9987
> Bug: webrtc:12445
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33204}

TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org

# Not skipping CQ checks because this is a reland.

Bug: webrtc:9987
Bug: webrtc:12445
Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33219}
2021-02-10 13:40:22 +00:00

169 lines
5.7 KiB
C++

/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/sctp_transport.h"
#include <algorithm>
#include <utility>
#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/synchronization/sequence_checker.h"
namespace webrtc {
SctpTransport::SctpTransport(
std::unique_ptr<cricket::SctpTransportInternal> internal)
: owner_thread_(rtc::Thread::Current()),
info_(SctpTransportState::kNew),
internal_sctp_transport_(std::move(internal)) {
RTC_DCHECK(internal_sctp_transport_.get());
internal_sctp_transport_->SignalAssociationChangeCommunicationUp.connect(
this, &SctpTransport::OnAssociationChangeCommunicationUp);
// TODO(https://bugs.webrtc.org/10360): Add handlers for transport closing.
if (dtls_transport_) {
UpdateInformation(SctpTransportState::kConnecting);
} else {
UpdateInformation(SctpTransportState::kNew);
}
}
SctpTransport::~SctpTransport() {
// We depend on the network thread to call Clear() before dropping
// its last reference to this object.
RTC_DCHECK(owner_thread_->IsCurrent() || !internal_sctp_transport_);
}
SctpTransportInformation SctpTransport::Information() const {
// TODO(tommi): Update PeerConnection::GetSctpTransport to hand out a proxy
// to the transport so that we can be sure that methods get called on the
// expected thread. Chromium currently calls this method from
// TransceiverStateSurfacer.
if (!owner_thread_->IsCurrent()) {
return owner_thread_->Invoke<SctpTransportInformation>(
RTC_FROM_HERE, [this] { return Information(); });
}
RTC_DCHECK_RUN_ON(owner_thread_);
return info_;
}
void SctpTransport::RegisterObserver(SctpTransportObserverInterface* observer) {
RTC_DCHECK_RUN_ON(owner_thread_);
RTC_DCHECK(observer);
RTC_DCHECK(!observer_);
observer_ = observer;
}
void SctpTransport::UnregisterObserver() {
RTC_DCHECK_RUN_ON(owner_thread_);
observer_ = nullptr;
}
rtc::scoped_refptr<DtlsTransportInterface> SctpTransport::dtls_transport()
const {
RTC_DCHECK_RUN_ON(owner_thread_);
return dtls_transport_;
}
// Internal functions
void SctpTransport::Clear() {
RTC_DCHECK_RUN_ON(owner_thread_);
RTC_DCHECK(internal());
// Note that we delete internal_sctp_transport_, but
// only drop the reference to dtls_transport_.
dtls_transport_ = nullptr;
internal_sctp_transport_ = nullptr;
UpdateInformation(SctpTransportState::kClosed);
}
void SctpTransport::SetDtlsTransport(
rtc::scoped_refptr<DtlsTransport> transport) {
RTC_DCHECK_RUN_ON(owner_thread_);
SctpTransportState next_state = info_.state();
dtls_transport_ = transport;
if (internal_sctp_transport_) {
if (transport) {
internal_sctp_transport_->SetDtlsTransport(transport->internal());
transport->internal()->SignalDtlsState.connect(
this, &SctpTransport::OnDtlsStateChange);
if (info_.state() == SctpTransportState::kNew) {
next_state = SctpTransportState::kConnecting;
}
} else {
internal_sctp_transport_->SetDtlsTransport(nullptr);
}
}
UpdateInformation(next_state);
}
void SctpTransport::Start(int local_port,
int remote_port,
int max_message_size) {
RTC_DCHECK_RUN_ON(owner_thread_);
info_ = SctpTransportInformation(info_.state(), info_.dtls_transport(),
max_message_size, info_.MaxChannels());
if (!internal()->Start(local_port, remote_port, max_message_size)) {
RTC_LOG(LS_ERROR) << "Failed to push down SCTP parameters, closing.";
UpdateInformation(SctpTransportState::kClosed);
}
}
void SctpTransport::UpdateInformation(SctpTransportState state) {
RTC_DCHECK_RUN_ON(owner_thread_);
bool must_send_update = (state != info_.state());
// TODO(https://bugs.webrtc.org/10358): Update max channels from internal
// SCTP transport when available.
if (internal_sctp_transport_) {
info_ = SctpTransportInformation(
state, dtls_transport_, info_.MaxMessageSize(), info_.MaxChannels());
} else {
info_ = SctpTransportInformation(
state, dtls_transport_, info_.MaxMessageSize(), info_.MaxChannels());
}
if (observer_ && must_send_update) {
observer_->OnStateChange(info_);
}
}
void SctpTransport::OnAssociationChangeCommunicationUp() {
RTC_DCHECK_RUN_ON(owner_thread_);
RTC_DCHECK(internal_sctp_transport_);
if (internal_sctp_transport_->max_outbound_streams() &&
internal_sctp_transport_->max_inbound_streams()) {
int max_channels =
std::min(*(internal_sctp_transport_->max_outbound_streams()),
*(internal_sctp_transport_->max_inbound_streams()));
// Record max channels.
info_ = SctpTransportInformation(info_.state(), info_.dtls_transport(),
info_.MaxMessageSize(), max_channels);
}
UpdateInformation(SctpTransportState::kConnected);
}
void SctpTransport::OnDtlsStateChange(cricket::DtlsTransportInternal* transport,
cricket::DtlsTransportState state) {
RTC_DCHECK_RUN_ON(owner_thread_);
RTC_CHECK(transport == dtls_transport_->internal());
if (state == cricket::DTLS_TRANSPORT_CLOSED ||
state == cricket::DTLS_TRANSPORT_FAILED) {
UpdateInformation(SctpTransportState::kClosed);
// TODO(http://bugs.webrtc.org/11090): Close all the data channels
}
}
} // namespace webrtc