6f68254ac3
This code looked a bit weird before this CL - probably because of old refactorings. In JsepTransport constructor, there is a DCHECK assuring that the RTP DTLS transport is always present, so it can be passed directly to the SctpTransport constructor, which avoids having the SetDtlsTransport method in it. Also, in the SctpTransport constructor, there was code that would set the SCTP transport state to `kConnecting` if the DTLS transport was present, but that was dead code, as it was always `nullptr` inside the constructor before this CL. With this CL, it's always present, and the SCTP Transport's state will initially always be `kConnecting` now. Which is a step to deprecating the `kNew` state that doesn't exist in https://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate. One test case was modified, as it didn't test the reality. The test created a SctpTransport, registered an observer, and added the DTLS transport, and expected to receive a "statechange" from `kNew` (which is not a state that exists in the spec) to `kConnecting`. If the test had tested the opposite ordering - adding the DTLS transport first, and then adding an observer, it wouldn't have experienced this. And since in reality (with the implementation of JsepTransport before and after this CL), it always adds the DTLS transport before any observer is registered. So it wouldn't ever be fired, outside of tests. Bug: webrtc:15897 Change-Id: I6ac24e0a331b686eb400fcf388ece50f2ad46a32 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345420 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41987}
104 lines
3.9 KiB
C++
104 lines
3.9 KiB
C++
/*
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* Copyright 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_SCTP_TRANSPORT_H_
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#define PC_SCTP_TRANSPORT_H_
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#include <memory>
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#include "api/dtls_transport_interface.h"
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#include "api/scoped_refptr.h"
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#include "api/sctp_transport_interface.h"
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#include "api/sequence_checker.h"
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#include "api/transport/data_channel_transport_interface.h"
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#include "media/sctp/sctp_transport_internal.h"
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#include "p2p/base/dtls_transport_internal.h"
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#include "pc/dtls_transport.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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// This implementation wraps a cricket::SctpTransport, and takes
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// ownership of it.
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// This object must be constructed and updated on the networking thread,
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// the same thread as the one the cricket::SctpTransportInternal object
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// lives on.
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class SctpTransport : public SctpTransportInterface,
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public DataChannelTransportInterface {
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public:
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SctpTransport(std::unique_ptr<cricket::SctpTransportInternal> internal,
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rtc::scoped_refptr<DtlsTransport> dtls_transport);
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// SctpTransportInterface
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override;
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SctpTransportInformation Information() const override;
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void RegisterObserver(SctpTransportObserverInterface* observer) override;
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void UnregisterObserver() override;
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// DataChannelTransportInterface
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RTCError OpenChannel(int channel_id) override;
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RTCError SendData(int channel_id,
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const SendDataParams& params,
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const rtc::CopyOnWriteBuffer& buffer) override;
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RTCError CloseChannel(int channel_id) override;
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void SetDataSink(DataChannelSink* sink) override;
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bool IsReadyToSend() const override;
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size_t buffered_amount(int channel_id) const override;
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size_t buffered_amount_low_threshold(int channel_id) const override;
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void SetBufferedAmountLowThreshold(int channel_id, size_t bytes) override;
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// Internal functions
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void Clear();
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// Initialize the cricket::SctpTransport. This can be called from
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// the signaling thread.
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void Start(int local_port, int remote_port, int max_message_size);
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// TODO(https://bugs.webrtc.org/10629): Move functions that need
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// internal() to be functions on the SctpTransport interface,
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// and make the internal() function private.
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cricket::SctpTransportInternal* internal() {
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RTC_DCHECK_RUN_ON(owner_thread_);
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return internal_sctp_transport_.get();
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}
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const cricket::SctpTransportInternal* internal() const {
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RTC_DCHECK_RUN_ON(owner_thread_);
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return internal_sctp_transport_.get();
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}
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protected:
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~SctpTransport() override;
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private:
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void UpdateInformation(SctpTransportState state);
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void OnInternalReadyToSendData();
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void OnAssociationChangeCommunicationUp();
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void OnInternalClosingProcedureStartedRemotely(int sid);
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void OnInternalClosingProcedureComplete(int sid);
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void OnDtlsStateChange(cricket::DtlsTransportInternal* transport,
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DtlsTransportState state);
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// NOTE: `owner_thread_` is the thread that the SctpTransport object is
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// constructed on. In the context of PeerConnection, it's the network thread.
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rtc::Thread* const owner_thread_;
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SctpTransportInformation info_ RTC_GUARDED_BY(owner_thread_);
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std::unique_ptr<cricket::SctpTransportInternal> internal_sctp_transport_
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RTC_GUARDED_BY(owner_thread_);
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SctpTransportObserverInterface* observer_ RTC_GUARDED_BY(owner_thread_) =
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nullptr;
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rtc::scoped_refptr<DtlsTransport> dtls_transport_
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RTC_GUARDED_BY(owner_thread_);
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};
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} // namespace webrtc
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#endif // PC_SCTP_TRANSPORT_H_
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