webrtc_m130/pc/data_channel_controller.h
Victor Boivie fea41f540c pc: Include SCTP queued bytes in buffered_amount
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.

Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.

This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.

This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.

Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
2024-03-13 15:44:17 +00:00

167 lines
6.7 KiB
C++

/*
* Copyright 2019 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.
*/
#ifndef PC_DATA_CHANNEL_CONTROLLER_H_
#define PC_DATA_CHANNEL_CONTROLLER_H_
#include <string>
#include <vector>
#include "api/data_channel_interface.h"
#include "api/rtc_error.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/data_channel_transport_interface.h"
#include "pc/data_channel_utils.h"
#include "pc/sctp_data_channel.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/weak_ptr.h"
namespace webrtc {
class PeerConnectionInternal;
class DataChannelController : public SctpDataChannelControllerInterface,
public DataChannelSink {
public:
explicit DataChannelController(PeerConnectionInternal* pc) : pc_(pc) {}
~DataChannelController();
// Not copyable or movable.
DataChannelController(DataChannelController&) = delete;
DataChannelController& operator=(const DataChannelController& other) = delete;
DataChannelController(DataChannelController&&) = delete;
DataChannelController& operator=(DataChannelController&& other) = delete;
// Implements
// SctpDataChannelProviderInterface.
RTCError SendData(StreamId sid,
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload) override;
void AddSctpDataStream(StreamId sid) override;
void RemoveSctpDataStream(StreamId sid) override;
void OnChannelStateChanged(SctpDataChannel* channel,
DataChannelInterface::DataState state) override;
size_t buffered_amount(StreamId sid) const override;
// Implements DataChannelSink.
void OnDataReceived(int channel_id,
DataMessageType type,
const rtc::CopyOnWriteBuffer& buffer) override;
void OnChannelClosing(int channel_id) override;
void OnChannelClosed(int channel_id) override;
void OnReadyToSend() override;
void OnTransportClosed(RTCError error) override;
// Called as part of destroying the owning PeerConnection.
void PrepareForShutdown();
// Called from PeerConnection::SetupDataChannelTransport_n
void SetupDataChannelTransport_n(DataChannelTransportInterface* transport);
// Called from PeerConnection::TeardownDataChannelTransport_n
void TeardownDataChannelTransport_n(RTCError error);
// Called from PeerConnection::OnTransportChanged
// to make required changes to datachannels' transports.
void OnTransportChanged(
DataChannelTransportInterface* data_channel_transport);
// Called from PeerConnection::GetDataChannelStats on the signaling thread.
std::vector<DataChannelStats> GetDataChannelStats() const;
// Creates channel and adds it to the collection of DataChannels that will
// be offered in a SessionDescription, and wraps it in a proxy object.
RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>>
InternalCreateDataChannelWithProxy(const std::string& label,
const InternalDataChannelInit& config);
void AllocateSctpSids(rtc::SSLRole role);
// Check if data channels are currently tracked. Used to decide whether a
// rejected m=application section should be reoffered.
bool HasDataChannels() const;
// At some point in time, a data channel has existed.
bool HasUsedDataChannels() const;
protected:
rtc::Thread* network_thread() const;
rtc::Thread* signaling_thread() const;
private:
void OnSctpDataChannelClosed(SctpDataChannel* channel);
// Creates a new SctpDataChannel object on the network thread.
RTCErrorOr<rtc::scoped_refptr<SctpDataChannel>> CreateDataChannel(
const std::string& label,
InternalDataChannelInit& config) RTC_RUN_ON(network_thread());
// Parses and handles open messages. Returns true if the message is an open
// message and should be considered to be handled, false otherwise.
bool HandleOpenMessage_n(int channel_id,
DataMessageType type,
const rtc::CopyOnWriteBuffer& buffer)
RTC_RUN_ON(network_thread());
// Called when a valid data channel OPEN message is received.
void OnDataChannelOpenMessage(rtc::scoped_refptr<SctpDataChannel> channel,
bool ready_to_send)
RTC_RUN_ON(signaling_thread());
// Accepts a `StreamId` which may be pre-negotiated or unassigned. For
// pre-negotiated sids, attempts to reserve the sid in the allocation pool,
// for unassigned sids attempts to generate a new sid if possible. Returns
// RTCError::OK() if the sid is reserved (and may have been generated) or
// if not enough information exists to generate a sid, in which case the sid
// will still be unassigned upon return, but will be assigned later.
// If the pool has been exhausted or a sid has already been reserved, an
// error will be returned.
RTCError ReserveOrAllocateSid(absl::optional<StreamId>& sid,
absl::optional<rtc::SSLRole> fallback_ssl_role)
RTC_RUN_ON(network_thread());
// Called when all data channels need to be notified of a transport channel
// (calls OnTransportChannelCreated on the signaling thread).
void NotifyDataChannelsOfTransportCreated();
void set_data_channel_transport(DataChannelTransportInterface* transport);
// Plugin transport used for data channels. Pointer may be accessed and
// checked from any thread, but the object may only be touched on the
// network thread.
DataChannelTransportInterface* data_channel_transport_
RTC_GUARDED_BY(network_thread()) = nullptr;
SctpSidAllocator sid_allocator_ RTC_GUARDED_BY(network_thread());
std::vector<rtc::scoped_refptr<SctpDataChannel>> sctp_data_channels_n_
RTC_GUARDED_BY(network_thread());
enum class DataChannelUsage : uint8_t {
kNeverUsed = 0,
kHaveBeenUsed,
kInUse
};
DataChannelUsage channel_usage_ RTC_GUARDED_BY(signaling_thread()) =
DataChannelUsage::kNeverUsed;
// Owning PeerConnection.
PeerConnectionInternal* const pc_;
// The weak pointers must be dereferenced and invalidated on the network
// thread only.
rtc::WeakPtrFactory<DataChannelController> weak_factory_
RTC_GUARDED_BY(network_thread()){this};
ScopedTaskSafety signaling_safety_;
};
} // namespace webrtc
#endif // PC_DATA_CHANNEL_CONTROLLER_H_