Remove SctpDataChannel::writable_
This flag isn't needed for sctp data channels. Bug: none Change-Id: I07b8ba2c5186729b8a5edb4d2bba7b800335ab5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299074 Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39701}
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@ -117,10 +117,6 @@ bool InternalDataChannelInit::IsValid() const {
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return true;
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}
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SctpSidAllocator::SctpSidAllocator() {
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sequence_checker_.Detach();
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}
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StreamId SctpSidAllocator::AllocateSid(rtc::SSLRole role) {
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RTC_DCHECK_RUN_ON(&sequence_checker_);
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int potential_sid = (role == rtc::SSL_CLIENT) ? 0 : 1;
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@ -389,9 +385,7 @@ void SctpDataChannel::OnTransportChannelCreated() {
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connected_to_transport_ = true;
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// The sid may have been unassigned when controller_->ConnectDataChannel was
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// done. So always add the streams even if connected_to_transport_ is true.
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if (id_.HasValue() && connected_to_transport_) {
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if (id_.HasValue()) {
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network_thread_->BlockingCall(
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[c = controller_.get(), sid = id_] { c->AddSctpDataStream(sid); });
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}
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@ -475,9 +469,7 @@ void SctpDataChannel::OnDataReceived(DataMessageType type,
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void SctpDataChannel::OnTransportReady() {
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RTC_DCHECK_RUN_ON(signaling_thread_);
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// TODO(tommi, hta): We don't need the `writable_` flag for SCTP datachannels.
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// Remove it and just rely on `connected_to_transport_` instead.
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// In practice the transport is configured inside
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// TODO(bugs.webrtc.org/11547): The transport is configured inside
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// `PeerConnection::SetupDataChannelTransport_n`, which results in
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// `SctpDataChannel` getting the OnTransportChannelCreated callback, and then
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// that's immediately followed by calling `transport->SetDataSink` which is
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@ -488,7 +480,6 @@ void SctpDataChannel::OnTransportReady() {
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// be on for the below `Send*` calls, which currently do a BlockingCall
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// from the signaling thread to the network thread.
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RTC_DCHECK(connected_to_transport_);
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writable_ = true;
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SendQueuedControlMessages();
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SendQueuedDataMessages();
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@ -544,8 +535,8 @@ void SctpDataChannel::UpdateState() {
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WriteDataChannelOpenAckMessage(&payload);
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SendControlMessage(payload);
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}
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if (writable_ && (handshake_state_ == kHandshakeReady ||
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handshake_state_ == kHandshakeWaitingForAck)) {
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if (handshake_state_ == kHandshakeReady ||
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handshake_state_ == kHandshakeWaitingForAck) {
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SetState(kOpen);
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// If we have received buffers before the channel got writable.
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// Deliver them now.
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@ -717,7 +708,6 @@ void SctpDataChannel::QueueControlMessage(
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bool SctpDataChannel::SendControlMessage(const rtc::CopyOnWriteBuffer& buffer) {
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RTC_DCHECK_RUN_ON(signaling_thread_);
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RTC_DCHECK(writable_);
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RTC_DCHECK(connected_to_transport_);
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RTC_DCHECK(id_.HasValue());
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@ -86,7 +86,7 @@ struct InternalDataChannelInit : public DataChannelInit {
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// Helper class to allocate unique IDs for SCTP DataChannels.
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class SctpSidAllocator {
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public:
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SctpSidAllocator();
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SctpSidAllocator() = default;
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// Gets the first unused odd/even id based on the DTLS role. If `role` is
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// SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
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// otherwise, the id starts from 1 and takes odd numbers.
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@ -101,7 +101,8 @@ class SctpSidAllocator {
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private:
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flat_set<StreamId> used_sids_ RTC_GUARDED_BY(&sequence_checker_);
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RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
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RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_{
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SequenceChecker::kDetached};
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};
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// SctpDataChannel is an implementation of the DataChannelInterface based on
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@ -278,7 +279,6 @@ class SctpDataChannel : public DataChannelInterface {
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HandshakeState handshake_state_ RTC_GUARDED_BY(signaling_thread_) =
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kHandshakeInit;
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bool connected_to_transport_ RTC_GUARDED_BY(signaling_thread_) = false;
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bool writable_ RTC_GUARDED_BY(signaling_thread_) = false;
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// Did we already start the graceful SCTP closing procedure?
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bool started_closing_procedure_ RTC_GUARDED_BY(signaling_thread_) = false;
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// Control messages that always have to get sent out before any queued
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