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Side by Side Diff: talk/app/webrtc/datachannel.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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1 /*
2 * libjingle
3 * Copyright 2012 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28 #ifndef TALK_APP_WEBRTC_DATACHANNEL_H_
29 #define TALK_APP_WEBRTC_DATACHANNEL_H_
30
31 #include <deque>
32 #include <set>
33 #include <string>
34
35 #include "talk/app/webrtc/datachannelinterface.h"
36 #include "talk/app/webrtc/proxy.h"
37 #include "talk/session/media/channel.h"
38 #include "webrtc/base/messagehandler.h"
39 #include "webrtc/base/scoped_ref_ptr.h"
40 #include "webrtc/base/sigslot.h"
41 #include "webrtc/media/base/mediachannel.h"
42
43 namespace webrtc {
44
45 class DataChannel;
46
47 class DataChannelProviderInterface {
48 public:
49 // Sends the data to the transport.
50 virtual bool SendData(const cricket::SendDataParams& params,
51 const rtc::Buffer& payload,
52 cricket::SendDataResult* result) = 0;
53 // Connects to the transport signals.
54 virtual bool ConnectDataChannel(DataChannel* data_channel) = 0;
55 // Disconnects from the transport signals.
56 virtual void DisconnectDataChannel(DataChannel* data_channel) = 0;
57 // Adds the data channel SID to the transport for SCTP.
58 virtual void AddSctpDataStream(int sid) = 0;
59 // Removes the data channel SID from the transport for SCTP.
60 virtual void RemoveSctpDataStream(int sid) = 0;
61 // Returns true if the transport channel is ready to send data.
62 virtual bool ReadyToSendData() const = 0;
63
64 protected:
65 virtual ~DataChannelProviderInterface() {}
66 };
67
68 struct InternalDataChannelInit : public DataChannelInit {
69 enum OpenHandshakeRole {
70 kOpener,
71 kAcker,
72 kNone
73 };
74 // The default role is kOpener because the default |negotiated| is false.
75 InternalDataChannelInit() : open_handshake_role(kOpener) {}
76 explicit InternalDataChannelInit(const DataChannelInit& base)
77 : DataChannelInit(base), open_handshake_role(kOpener) {
78 // If the channel is externally negotiated, do not send the OPEN message.
79 if (base.negotiated) {
80 open_handshake_role = kNone;
81 }
82 }
83
84 OpenHandshakeRole open_handshake_role;
85 };
86
87 // Helper class to allocate unique IDs for SCTP DataChannels
88 class SctpSidAllocator {
89 public:
90 // Gets the first unused odd/even id based on the DTLS role. If |role| is
91 // SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
92 // otherwise, the id starts from 1 and takes odd numbers.
93 // Returns false if no id can be allocated.
94 bool AllocateSid(rtc::SSLRole role, int* sid);
95
96 // Attempts to reserve a specific sid. Returns false if it's unavailable.
97 bool ReserveSid(int sid);
98
99 // Indicates that |sid| isn't in use any more, and is thus available again.
100 void ReleaseSid(int sid);
101
102 private:
103 // Checks if |sid| is available to be assigned to a new SCTP data channel.
104 bool IsSidAvailable(int sid) const;
105
106 std::set<int> used_sids_;
107 };
108
109 // DataChannel is a an implementation of the DataChannelInterface based on
110 // libjingle's data engine. It provides an implementation of unreliable or
111 // reliabledata channels. Currently this class is specifically designed to use
112 // both RtpDataEngine and SctpDataEngine.
113
114 // DataChannel states:
115 // kConnecting: The channel has been created the transport might not yet be
116 // ready.
117 // kOpen: The channel have a local SSRC set by a call to UpdateSendSsrc
118 // and a remote SSRC set by call to UpdateReceiveSsrc and the transport
119 // has been writable once.
120 // kClosing: DataChannelInterface::Close has been called or UpdateReceiveSsrc
121 // has been called with SSRC==0
122 // kClosed: Both UpdateReceiveSsrc and UpdateSendSsrc has been called with
123 // SSRC==0.
124 class DataChannel : public DataChannelInterface,
125 public sigslot::has_slots<>,
126 public rtc::MessageHandler {
127 public:
128 static rtc::scoped_refptr<DataChannel> Create(
129 DataChannelProviderInterface* provider,
130 cricket::DataChannelType dct,
131 const std::string& label,
132 const InternalDataChannelInit& config);
133
134 virtual void RegisterObserver(DataChannelObserver* observer);
135 virtual void UnregisterObserver();
136
137 virtual std::string label() const { return label_; }
138 virtual bool reliable() const;
139 virtual bool ordered() const { return config_.ordered; }
140 virtual uint16_t maxRetransmitTime() const {
141 return config_.maxRetransmitTime;
142 }
143 virtual uint16_t maxRetransmits() const { return config_.maxRetransmits; }
144 virtual std::string protocol() const { return config_.protocol; }
145 virtual bool negotiated() const { return config_.negotiated; }
146 virtual int id() const { return config_.id; }
147 virtual uint64_t buffered_amount() const;
148 virtual void Close();
149 virtual DataState state() const { return state_; }
150 virtual bool Send(const DataBuffer& buffer);
151
152 // rtc::MessageHandler override.
153 virtual void OnMessage(rtc::Message* msg);
154
155 // Called when the channel's ready to use. That can happen when the
156 // underlying DataMediaChannel becomes ready, or when this channel is a new
157 // stream on an existing DataMediaChannel, and we've finished negotiation.
158 void OnChannelReady(bool writable);
159
160 // Sigslots from cricket::DataChannel
161 void OnDataReceived(cricket::DataChannel* channel,
162 const cricket::ReceiveDataParams& params,
163 const rtc::Buffer& payload);
164 void OnStreamClosedRemotely(uint32_t sid);
165
166 // The remote peer request that this channel should be closed.
167 void RemotePeerRequestClose();
168
169 // The following methods are for SCTP only.
170
171 // Sets the SCTP sid and adds to transport layer if not set yet. Should only
172 // be called once.
173 void SetSctpSid(int sid);
174 // Called when the transport channel is created.
175 // Only needs to be called for SCTP data channels.
176 void OnTransportChannelCreated();
177 // Called when the transport channel is destroyed.
178 void OnTransportChannelDestroyed();
179
180 // The following methods are for RTP only.
181
182 // Set the SSRC this channel should use to send data on the
183 // underlying data engine. |send_ssrc| == 0 means that the channel is no
184 // longer part of the session negotiation.
185 void SetSendSsrc(uint32_t send_ssrc);
186 // Set the SSRC this channel should use to receive data from the
187 // underlying data engine.
188 void SetReceiveSsrc(uint32_t receive_ssrc);
189
190 cricket::DataChannelType data_channel_type() const {
191 return data_channel_type_;
192 }
193
194 // Emitted when state transitions to kClosed.
195 // In the case of SCTP channels, this signal can be used to tell when the
196 // channel's sid is free.
197 sigslot::signal1<DataChannel*> SignalClosed;
198
199 protected:
200 DataChannel(DataChannelProviderInterface* client,
201 cricket::DataChannelType dct,
202 const std::string& label);
203 virtual ~DataChannel();
204
205 private:
206 // A packet queue which tracks the total queued bytes. Queued packets are
207 // owned by this class.
208 class PacketQueue {
209 public:
210 PacketQueue();
211 ~PacketQueue();
212
213 size_t byte_count() const {
214 return byte_count_;
215 }
216
217 bool Empty() const;
218
219 DataBuffer* Front();
220
221 void Pop();
222
223 void Push(DataBuffer* packet);
224
225 void Clear();
226
227 void Swap(PacketQueue* other);
228
229 private:
230 std::deque<DataBuffer*> packets_;
231 size_t byte_count_;
232 };
233
234 // The OPEN(_ACK) signaling state.
235 enum HandshakeState {
236 kHandshakeInit,
237 kHandshakeShouldSendOpen,
238 kHandshakeShouldSendAck,
239 kHandshakeWaitingForAck,
240 kHandshakeReady
241 };
242
243 bool Init(const InternalDataChannelInit& config);
244 void DoClose();
245 void UpdateState();
246 void SetState(DataState state);
247 void DisconnectFromProvider();
248
249 void DeliverQueuedReceivedData();
250
251 void SendQueuedDataMessages();
252 bool SendDataMessage(const DataBuffer& buffer, bool queue_if_blocked);
253 bool QueueSendDataMessage(const DataBuffer& buffer);
254
255 void SendQueuedControlMessages();
256 void QueueControlMessage(const rtc::Buffer& buffer);
257 bool SendControlMessage(const rtc::Buffer& buffer);
258
259 std::string label_;
260 InternalDataChannelInit config_;
261 DataChannelObserver* observer_;
262 DataState state_;
263 cricket::DataChannelType data_channel_type_;
264 DataChannelProviderInterface* provider_;
265 HandshakeState handshake_state_;
266 bool connected_to_provider_;
267 bool send_ssrc_set_;
268 bool receive_ssrc_set_;
269 bool writable_;
270 uint32_t send_ssrc_;
271 uint32_t receive_ssrc_;
272 // Control messages that always have to get sent out before any queued
273 // data.
274 PacketQueue queued_control_data_;
275 PacketQueue queued_received_data_;
276 PacketQueue queued_send_data_;
277 };
278
279 // Define proxy for DataChannelInterface.
280 BEGIN_PROXY_MAP(DataChannel)
281 PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*)
282 PROXY_METHOD0(void, UnregisterObserver)
283 PROXY_CONSTMETHOD0(std::string, label)
284 PROXY_CONSTMETHOD0(bool, reliable)
285 PROXY_CONSTMETHOD0(bool, ordered)
286 PROXY_CONSTMETHOD0(uint16_t, maxRetransmitTime)
287 PROXY_CONSTMETHOD0(uint16_t, maxRetransmits)
288 PROXY_CONSTMETHOD0(std::string, protocol)
289 PROXY_CONSTMETHOD0(bool, negotiated)
290 PROXY_CONSTMETHOD0(int, id)
291 PROXY_CONSTMETHOD0(DataState, state)
292 PROXY_CONSTMETHOD0(uint64_t, buffered_amount)
293 PROXY_METHOD0(void, Close)
294 PROXY_METHOD1(bool, Send, const DataBuffer&)
295 END_PROXY()
296
297 } // namespace webrtc
298
299 #endif // TALK_APP_WEBRTC_DATACHANNEL_H_
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