Revert of Separating SCTP code from BaseChannel/MediaChannel.

webrtc/media/sctp/sctptransportinternal.h

Index: webrtc/media/sctp/sctptransportinternal.h
diff --git a/webrtc/media/sctp/sctptransportinternal.h b/webrtc/media/sctp/sctptransportinternal.h
deleted file mode 100644
index 7dd6bc7ea7a8f93b7c77c16b08f9b8cc2ccfc5af..0000000000000000000000000000000000000000
--- a/webrtc/media/sctp/sctptransportinternal.h
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- *  Copyright (c) 2016 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 WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_
-#define WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_
-
-// TODO(deadbeef): Move SCTP code out of media/, and make it not depend on
-// anything in media/.
-
-#include <memory>  // for unique_ptr
-#include <string>
-#include <vector>
-
-#include "webrtc/base/copyonwritebuffer.h"
-#include "webrtc/base/thread.h"
-// For SendDataParams/ReceiveDataParams.
-// TODO(deadbeef): Use something else for SCTP. It's confusing that we use an
-// SSRC field for SID.
-#include "webrtc/media/base/mediachannel.h"
-#include "webrtc/p2p/base/transportchannel.h"
-
-namespace cricket {
-
-// The number of outgoing streams that we'll negotiate. Since stream IDs (SIDs)
-// are 0-based, the highest usable SID is 1023.
-//
-// It's recommended to use the maximum of 65535 in:
-// https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2
-// However, we use 1024 in order to save memory. usrsctp allocates 104 bytes
-// for each pair of incoming/outgoing streams (on a 64-bit system), so 65535
-// streams would waste ~6MB.
-//
-// Note: "max" and "min" here are inclusive.
-constexpr uint16_t kMaxSctpStreams = 1024;
-constexpr uint16_t kMaxSctpSid = kMaxSctpStreams - 1;
-constexpr uint16_t kMinSctpSid = 0;
-
-// This is the default SCTP port to use. It is passed along the wire and the
-// connectee and connector must be using the same port. It is not related to the
-// ports at the IP level. (Corresponds to: sockaddr_conn.sconn_port in
-// usrsctp.h)
-const int kSctpDefaultPort = 5000;
-
-// Abstract SctpTransport interface for use internally (by
-// PeerConnection/WebRtcSession/etc.). Exists to allow mock/fake SctpTransports
-// to be created.
-class SctpTransportInternal {
- public:
-  virtual ~SctpTransportInternal() {}
-
-  // Changes what underlying DTLS channel is uses. Used when switching which
-  // bundled transport the SctpTransport uses.
-  // Assumes |channel| is non-null.
-  virtual void SetTransportChannel(TransportChannel* channel) = 0;
-
-  // When Start is called, connects as soon as possible; this can be called
-  // before DTLS completes, in which case the connection will begin when DTLS
-  // completes. This method can be called multiple times, though not if either
-  // of the ports are changed.
-  //
-  // |local_sctp_port| and |remote_sctp_port| are passed along the wire and the
-  // listener and connector must be using the same port. They are not related
-  // to the ports at the IP level. If set to -1, we default to
-  // kSctpDefaultPort.
-  //
-  // TODO(deadbeef): Add remote max message size as parameter to Start, once we
-  // start supporting it.
-  // TODO(deadbeef): Support calling Start with different local/remote ports
-  // and create a new association? Not clear if this is something we need to
-  // support though. See: https://github.com/w3c/webrtc-pc/issues/979
-  virtual bool Start(int local_sctp_port, int remote_sctp_port) = 0;
-
-  // NOTE: Initially there was a "Stop" method here, but it was never used, so
-  // it was removed.
-
-  // Informs SctpTransport that |sid| will start being used. Returns false if
-  // it is impossible to use |sid|, or if it's already in use.
-  // Until calling this, can't send data using |sid|.
-  // TODO(deadbeef): Actually implement the "returns false if |sid| can't be
-  // used" part. See:
-  // https://bugs.chromium.org/p/chromium/issues/detail?id=619849
-  virtual bool OpenStream(int sid) = 0;
-  // The inverse of OpenStream. When this method returns, the reset process may
-  // have not finished but it will have begun.
-  // TODO(deadbeef): We need a way to tell when it's done. See:
-  // https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
-  virtual bool ResetStream(int sid) = 0;
-  // Send data down this channel (will be wrapped as SCTP packets then given to
-  // usrsctp that will then post the network interface).
-  // Returns true iff successful data somewhere on the send-queue/network.
-  // Uses |params.ssrc| as the SCTP sid.
-  virtual bool SendData(const SendDataParams& params,
-                        const rtc::CopyOnWriteBuffer& payload,
-                        SendDataResult* result = nullptr) = 0;
-
-  // Indicates when the SCTP socket is created and not blocked by congestion
-  // control. This changes to false when SDR_BLOCK is returned from SendData,
-  // and
-  // changes to true when SignalReadyToSendData is fired. The underlying DTLS/
-  // ICE channels may be unwritable while ReadyToSendData is true, because data
-  // can still be queued in usrsctp.
-  virtual bool ReadyToSendData() = 0;
-
-  sigslot::signal0<> SignalReadyToSendData;
-  // ReceiveDataParams includes SID, seq num, timestamp, etc. CopyOnWriteBuffer
-  // contains message payload.
-  sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&>
-      SignalDataReceived;
-  // Parameter is SID of closed stream.
-  sigslot::signal1<int> SignalStreamClosedRemotely;
-
-  // Helper for debugging.
-  virtual void set_debug_name_for_testing(const char* debug_name) = 0;
-};
-
-// Factory class which can be used to allow fake SctpTransports to be injected
-// for testing. Or, theoretically, SctpTransportInternal implementations that
-// use something other than usrsctp.
-class SctpTransportInternalFactory {
- public:
-  virtual ~SctpTransportInternalFactory() {}
-
-  // Create an SCTP transport using |channel| for the underlying transport.
-  virtual std::unique_ptr<SctpTransportInternal> CreateSctpTransport(
-      TransportChannel* channel) = 0;
-};
-
-}  // namespace cricket
-
-#endif  // WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_