|  | /* | 
|  | *  Copyright 2020 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_RTP_DATA_CHANNEL_H_ | 
|  | #define PC_RTP_DATA_CHANNEL_H_ | 
|  |  | 
|  | #include <stdint.h> | 
|  |  | 
|  | #include <memory> | 
|  | #include <string> | 
|  |  | 
|  | #include "absl/types/optional.h" | 
|  | #include "api/data_channel_interface.h" | 
|  | #include "api/priority.h" | 
|  | #include "api/rtc_error.h" | 
|  | #include "api/scoped_refptr.h" | 
|  | #include "api/transport/data_channel_transport_interface.h" | 
|  | #include "media/base/media_channel.h" | 
|  | #include "pc/channel.h" | 
|  | #include "pc/data_channel_utils.h" | 
|  | #include "rtc_base/async_invoker.h" | 
|  | #include "rtc_base/copy_on_write_buffer.h" | 
|  | #include "rtc_base/third_party/sigslot/sigslot.h" | 
|  | #include "rtc_base/thread.h" | 
|  | #include "rtc_base/thread_annotations.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class RtpDataChannel; | 
|  |  | 
|  | // TODO(deadbeef): Once RTP data channels go away, get rid of this and have | 
|  | // DataChannel depend on SctpTransportInternal (pure virtual SctpTransport | 
|  | // interface) instead. | 
|  | class RtpDataChannelProviderInterface { | 
|  | public: | 
|  | // Sends the data to the transport. | 
|  | virtual bool SendData(const cricket::SendDataParams& params, | 
|  | const rtc::CopyOnWriteBuffer& payload, | 
|  | cricket::SendDataResult* result) = 0; | 
|  | // Connects to the transport signals. | 
|  | virtual bool ConnectDataChannel(RtpDataChannel* data_channel) = 0; | 
|  | // Disconnects from the transport signals. | 
|  | virtual void DisconnectDataChannel(RtpDataChannel* data_channel) = 0; | 
|  | // Returns true if the transport channel is ready to send data. | 
|  | virtual bool ReadyToSendData() const = 0; | 
|  |  | 
|  | protected: | 
|  | virtual ~RtpDataChannelProviderInterface() {} | 
|  | }; | 
|  |  | 
|  | // RtpDataChannel is an implementation of the DataChannelInterface based on | 
|  | // libjingle's data engine. It provides an implementation of unreliable data | 
|  | // channels. | 
|  |  | 
|  | // DataChannel states: | 
|  | // kConnecting: The channel has been created the transport might not yet be | 
|  | //              ready. | 
|  | // kOpen: The channel have a local SSRC set by a call to UpdateSendSsrc | 
|  | //        and a remote SSRC set by call to UpdateReceiveSsrc and the transport | 
|  | //        has been writable once. | 
|  | // kClosing: DataChannelInterface::Close has been called or UpdateReceiveSsrc | 
|  | //           has been called with SSRC==0 | 
|  | // kClosed: Both UpdateReceiveSsrc and UpdateSendSsrc has been called with | 
|  | //          SSRC==0. | 
|  | class RtpDataChannel : public DataChannelInterface, | 
|  | public sigslot::has_slots<> { | 
|  | public: | 
|  | static rtc::scoped_refptr<RtpDataChannel> Create( | 
|  | RtpDataChannelProviderInterface* provider, | 
|  | const std::string& label, | 
|  | const DataChannelInit& config, | 
|  | rtc::Thread* signaling_thread); | 
|  |  | 
|  | // Instantiates an API proxy for a DataChannel instance that will be handed | 
|  | // out to external callers. | 
|  | static rtc::scoped_refptr<DataChannelInterface> CreateProxy( | 
|  | rtc::scoped_refptr<RtpDataChannel> channel); | 
|  |  | 
|  | void RegisterObserver(DataChannelObserver* observer) override; | 
|  | void UnregisterObserver() override; | 
|  |  | 
|  | std::string label() const override { return label_; } | 
|  | bool reliable() const override { return false; } | 
|  | bool ordered() const override { return config_.ordered; } | 
|  | // Backwards compatible accessors | 
|  | uint16_t maxRetransmitTime() const override { | 
|  | return config_.maxRetransmitTime ? *config_.maxRetransmitTime | 
|  | : static_cast<uint16_t>(-1); | 
|  | } | 
|  | uint16_t maxRetransmits() const override { | 
|  | return config_.maxRetransmits ? *config_.maxRetransmits | 
|  | : static_cast<uint16_t>(-1); | 
|  | } | 
|  | absl::optional<int> maxPacketLifeTime() const override { | 
|  | return config_.maxRetransmitTime; | 
|  | } | 
|  | absl::optional<int> maxRetransmitsOpt() const override { | 
|  | return config_.maxRetransmits; | 
|  | } | 
|  | std::string protocol() const override { return config_.protocol; } | 
|  | bool negotiated() const override { return config_.negotiated; } | 
|  | int id() const override { return config_.id; } | 
|  | Priority priority() const override { | 
|  | return config_.priority ? *config_.priority : Priority::kLow; | 
|  | } | 
|  |  | 
|  | virtual int internal_id() const { return internal_id_; } | 
|  |  | 
|  | uint64_t buffered_amount() const override { return 0; } | 
|  | void Close() override; | 
|  | DataState state() const override; | 
|  | RTCError error() const override; | 
|  | uint32_t messages_sent() const override; | 
|  | uint64_t bytes_sent() const override; | 
|  | uint32_t messages_received() const override; | 
|  | uint64_t bytes_received() const override; | 
|  | bool Send(const DataBuffer& buffer) override; | 
|  |  | 
|  | // Close immediately, ignoring any queued data or closing procedure. | 
|  | // This is called when SDP indicates a channel should be removed. | 
|  | void CloseAbruptlyWithError(RTCError error); | 
|  |  | 
|  | // Called when the channel's ready to use.  That can happen when the | 
|  | // underlying DataMediaChannel becomes ready, or when this channel is a new | 
|  | // stream on an existing DataMediaChannel, and we've finished negotiation. | 
|  | void OnChannelReady(bool writable); | 
|  |  | 
|  | // Slots for provider to connect signals to. | 
|  | void OnDataReceived(const cricket::ReceiveDataParams& params, | 
|  | const rtc::CopyOnWriteBuffer& payload); | 
|  |  | 
|  | // Called when the transport channel is unusable. | 
|  | // This method makes sure the DataChannel is disconnected and changes state | 
|  | // to kClosed. | 
|  | void OnTransportChannelClosed(); | 
|  |  | 
|  | DataChannelStats GetStats() const; | 
|  |  | 
|  | // The remote peer requested that this channel should be closed. | 
|  | void RemotePeerRequestClose(); | 
|  | // Set the SSRC this channel should use to send data on the | 
|  | // underlying data engine. |send_ssrc| == 0 means that the channel is no | 
|  | // longer part of the session negotiation. | 
|  | void SetSendSsrc(uint32_t send_ssrc); | 
|  | // Set the SSRC this channel should use to receive data from the | 
|  | // underlying data engine. | 
|  | void SetReceiveSsrc(uint32_t receive_ssrc); | 
|  |  | 
|  | // Emitted when state transitions to kOpen. | 
|  | sigslot::signal1<DataChannelInterface*> SignalOpened; | 
|  | // Emitted when state transitions to kClosed. | 
|  | sigslot::signal1<DataChannelInterface*> SignalClosed; | 
|  |  | 
|  | // Reset the allocator for internal ID values for testing, so that | 
|  | // the internal IDs generated are predictable. Test only. | 
|  | static void ResetInternalIdAllocatorForTesting(int new_value); | 
|  |  | 
|  | protected: | 
|  | RtpDataChannel(const DataChannelInit& config, | 
|  | RtpDataChannelProviderInterface* client, | 
|  | const std::string& label, | 
|  | rtc::Thread* signaling_thread); | 
|  | ~RtpDataChannel() override; | 
|  |  | 
|  | private: | 
|  | bool Init(); | 
|  | void UpdateState(); | 
|  | void SetState(DataState state); | 
|  | void DisconnectFromProvider(); | 
|  |  | 
|  | void DeliverQueuedReceivedData(); | 
|  |  | 
|  | bool SendDataMessage(const DataBuffer& buffer); | 
|  |  | 
|  | rtc::Thread* const signaling_thread_; | 
|  | const int internal_id_; | 
|  | const std::string label_; | 
|  | const DataChannelInit config_; | 
|  | DataChannelObserver* observer_ RTC_GUARDED_BY(signaling_thread_) = nullptr; | 
|  | DataState state_ RTC_GUARDED_BY(signaling_thread_) = kConnecting; | 
|  | RTCError error_ RTC_GUARDED_BY(signaling_thread_); | 
|  | uint32_t messages_sent_ RTC_GUARDED_BY(signaling_thread_) = 0; | 
|  | uint64_t bytes_sent_ RTC_GUARDED_BY(signaling_thread_) = 0; | 
|  | uint32_t messages_received_ RTC_GUARDED_BY(signaling_thread_) = 0; | 
|  | uint64_t bytes_received_ RTC_GUARDED_BY(signaling_thread_) = 0; | 
|  | RtpDataChannelProviderInterface* const provider_; | 
|  | bool connected_to_provider_ RTC_GUARDED_BY(signaling_thread_) = false; | 
|  | bool send_ssrc_set_ RTC_GUARDED_BY(signaling_thread_) = false; | 
|  | bool receive_ssrc_set_ RTC_GUARDED_BY(signaling_thread_) = false; | 
|  | bool writable_ RTC_GUARDED_BY(signaling_thread_) = false; | 
|  | uint32_t send_ssrc_ RTC_GUARDED_BY(signaling_thread_) = 0; | 
|  | uint32_t receive_ssrc_ RTC_GUARDED_BY(signaling_thread_) = 0; | 
|  | PacketQueue queued_received_data_ RTC_GUARDED_BY(signaling_thread_); | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // PC_RTP_DATA_CHANNEL_H_ |