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}
diff --git a/pc/sctp_data_channel.cc b/pc/sctp_data_channel.cc
index 2280174..caafa9d 100644
--- a/pc/sctp_data_channel.cc
+++ b/pc/sctp_data_channel.cc
@@ -117,10 +117,6 @@
return true;
}
-SctpSidAllocator::SctpSidAllocator() {
- sequence_checker_.Detach();
-}
-
StreamId SctpSidAllocator::AllocateSid(rtc::SSLRole role) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
int potential_sid = (role == rtc::SSL_CLIENT) ? 0 : 1;
@@ -389,9 +385,7 @@
connected_to_transport_ = true;
- // The sid may have been unassigned when controller_->ConnectDataChannel was
- // done. So always add the streams even if connected_to_transport_ is true.
- if (id_.HasValue() && connected_to_transport_) {
+ if (id_.HasValue()) {
network_thread_->BlockingCall(
[c = controller_.get(), sid = id_] { c->AddSctpDataStream(sid); });
}
@@ -475,9 +469,7 @@
void SctpDataChannel::OnTransportReady() {
RTC_DCHECK_RUN_ON(signaling_thread_);
- // TODO(tommi, hta): We don't need the `writable_` flag for SCTP datachannels.
- // Remove it and just rely on `connected_to_transport_` instead.
- // In practice the transport is configured inside
+ // TODO(bugs.webrtc.org/11547): The transport is configured inside
// `PeerConnection::SetupDataChannelTransport_n`, which results in
// `SctpDataChannel` getting the OnTransportChannelCreated callback, and then
// that's immediately followed by calling `transport->SetDataSink` which is
@@ -488,7 +480,6 @@
// be on for the below `Send*` calls, which currently do a BlockingCall
// from the signaling thread to the network thread.
RTC_DCHECK(connected_to_transport_);
- writable_ = true;
SendQueuedControlMessages();
SendQueuedDataMessages();
@@ -544,8 +535,8 @@
WriteDataChannelOpenAckMessage(&payload);
SendControlMessage(payload);
}
- if (writable_ && (handshake_state_ == kHandshakeReady ||
- handshake_state_ == kHandshakeWaitingForAck)) {
+ if (handshake_state_ == kHandshakeReady ||
+ handshake_state_ == kHandshakeWaitingForAck) {
SetState(kOpen);
// If we have received buffers before the channel got writable.
// Deliver them now.
@@ -717,7 +708,6 @@
bool SctpDataChannel::SendControlMessage(const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(signaling_thread_);
- RTC_DCHECK(writable_);
RTC_DCHECK(connected_to_transport_);
RTC_DCHECK(id_.HasValue());
diff --git a/pc/sctp_data_channel.h b/pc/sctp_data_channel.h
index 184603d..5cfa364 100644
--- a/pc/sctp_data_channel.h
+++ b/pc/sctp_data_channel.h
@@ -86,7 +86,7 @@
// Helper class to allocate unique IDs for SCTP DataChannels.
class SctpSidAllocator {
public:
- SctpSidAllocator();
+ SctpSidAllocator() = default;
// Gets the first unused odd/even id based on the DTLS role. If `role` is
// SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
// otherwise, the id starts from 1 and takes odd numbers.
@@ -101,7 +101,8 @@
private:
flat_set<StreamId> used_sids_ RTC_GUARDED_BY(&sequence_checker_);
- RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+ RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_{
+ SequenceChecker::kDetached};
};
// SctpDataChannel is an implementation of the DataChannelInterface based on
@@ -278,7 +279,6 @@
HandshakeState handshake_state_ RTC_GUARDED_BY(signaling_thread_) =
kHandshakeInit;
bool connected_to_transport_ RTC_GUARDED_BY(signaling_thread_) = false;
- bool writable_ RTC_GUARDED_BY(signaling_thread_) = false;
// Did we already start the graceful SCTP closing procedure?
bool started_closing_procedure_ RTC_GUARDED_BY(signaling_thread_) = false;
// Control messages that always have to get sent out before any queued
