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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2004 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_SESSION_MEDIA_CHANNEL_H_
29#define TALK_SESSION_MEDIA_CHANNEL_H_
30
31#include <string>
32#include <vector>
33
34#include "talk/base/asyncudpsocket.h"
35#include "talk/base/criticalsection.h"
36#include "talk/base/network.h"
37#include "talk/base/sigslot.h"
38#include "talk/base/window.h"
39#include "talk/media/base/mediachannel.h"
40#include "talk/media/base/mediaengine.h"
41#include "talk/media/base/screencastid.h"
42#include "talk/media/base/streamparams.h"
43#include "talk/media/base/videocapturer.h"
44#include "talk/p2p/base/session.h"
45#include "talk/p2p/client/socketmonitor.h"
46#include "talk/session/media/audiomonitor.h"
47#include "talk/session/media/mediamonitor.h"
48#include "talk/session/media/mediasession.h"
49#include "talk/session/media/rtcpmuxfilter.h"
50#include "talk/session/media/srtpfilter.h"
51#include "talk/session/media/ssrcmuxfilter.h"
52
53namespace cricket {
54
55struct CryptoParams;
56class MediaContentDescription;
57struct TypingMonitorOptions;
58class TypingMonitor;
59struct ViewRequest;
60
61enum SinkType {
62 SINK_PRE_CRYPTO, // Sink packets before encryption or after decryption.
63 SINK_POST_CRYPTO // Sink packets after encryption or before decryption.
64};
65
66// BaseChannel contains logic common to voice and video, including
67// enable/mute, marshaling calls to a worker thread, and
68// connection and media monitors.
wu@webrtc.org78187522013-10-07 23:32:02 +000069//
70// WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
71// This is required to avoid a data race between the destructor modifying the
72// vtable, and the media channel's thread using BaseChannel as the
73// NetworkInterface.
74
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075class BaseChannel
76 : public talk_base::MessageHandler, public sigslot::has_slots<>,
77 public MediaChannel::NetworkInterface {
78 public:
79 BaseChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine,
80 MediaChannel* channel, BaseSession* session,
81 const std::string& content_name, bool rtcp);
82 virtual ~BaseChannel();
83 bool Init(TransportChannel* transport_channel,
84 TransportChannel* rtcp_transport_channel);
wu@webrtc.org78187522013-10-07 23:32:02 +000085 // Deinit may be called multiple times and is simply ignored if it's alreay
86 // done.
87 void Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +000088
89 talk_base::Thread* worker_thread() const { return worker_thread_; }
90 BaseSession* session() const { return session_; }
91 const std::string& content_name() { return content_name_; }
92 TransportChannel* transport_channel() const {
93 return transport_channel_;
94 }
95 TransportChannel* rtcp_transport_channel() const {
96 return rtcp_transport_channel_;
97 }
98 bool enabled() const { return enabled_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +000099
100 // This function returns true if we are using SRTP.
101 bool secure() const { return srtp_filter_.IsActive(); }
102 // The following function returns true if we are using
103 // DTLS-based keying. If you turned off SRTP later, however
104 // you could have secure() == false and dtls_secure() == true.
105 bool secure_dtls() const { return dtls_keyed_; }
106 // This function returns true if we require secure channel for call setup.
107 bool secure_required() const { return secure_required_; }
108
109 bool writable() const { return writable_; }
110 bool IsStreamMuted(uint32 ssrc);
111
112 // Channel control
113 bool SetLocalContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000114 ContentAction action,
115 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000116 bool SetRemoteContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000117 ContentAction action,
118 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000119
120 bool Enable(bool enable);
121 // Mute sending media on the stream with SSRC |ssrc|
122 // If there is only one sending stream SSRC 0 can be used.
123 bool MuteStream(uint32 ssrc, bool mute);
124
125 // Multiplexing
126 bool AddRecvStream(const StreamParams& sp);
127 bool RemoveRecvStream(uint32 ssrc);
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000128 bool AddSendStream(const StreamParams& sp);
129 bool RemoveSendStream(uint32 ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000130
131 // Monitoring
132 void StartConnectionMonitor(int cms);
133 void StopConnectionMonitor();
134
135 void set_srtp_signal_silent_time(uint32 silent_time) {
136 srtp_filter_.set_signal_silent_time(silent_time);
137 }
138
139 void set_content_name(const std::string& content_name) {
140 ASSERT(signaling_thread()->IsCurrent());
141 ASSERT(!writable_);
142 if (session_->state() != BaseSession::STATE_INIT) {
143 LOG(LS_ERROR) << "Content name for a channel can be changed only "
144 << "when BaseSession is in STATE_INIT state.";
145 return;
146 }
147 content_name_ = content_name;
148 }
149
150 template <class T>
151 void RegisterSendSink(T* sink,
152 void (T::*OnPacket)(const void*, size_t, bool),
153 SinkType type) {
154 talk_base::CritScope cs(&signal_send_packet_cs_);
155 if (SINK_POST_CRYPTO == type) {
156 SignalSendPacketPostCrypto.disconnect(sink);
157 SignalSendPacketPostCrypto.connect(sink, OnPacket);
158 } else {
159 SignalSendPacketPreCrypto.disconnect(sink);
160 SignalSendPacketPreCrypto.connect(sink, OnPacket);
161 }
162 }
163
164 void UnregisterSendSink(sigslot::has_slots<>* sink,
165 SinkType type) {
166 talk_base::CritScope cs(&signal_send_packet_cs_);
167 if (SINK_POST_CRYPTO == type) {
168 SignalSendPacketPostCrypto.disconnect(sink);
169 } else {
170 SignalSendPacketPreCrypto.disconnect(sink);
171 }
172 }
173
174 bool HasSendSinks(SinkType type) {
175 talk_base::CritScope cs(&signal_send_packet_cs_);
176 if (SINK_POST_CRYPTO == type) {
177 return !SignalSendPacketPostCrypto.is_empty();
178 } else {
179 return !SignalSendPacketPreCrypto.is_empty();
180 }
181 }
182
183 template <class T>
184 void RegisterRecvSink(T* sink,
185 void (T::*OnPacket)(const void*, size_t, bool),
186 SinkType type) {
187 talk_base::CritScope cs(&signal_recv_packet_cs_);
188 if (SINK_POST_CRYPTO == type) {
189 SignalRecvPacketPostCrypto.disconnect(sink);
190 SignalRecvPacketPostCrypto.connect(sink, OnPacket);
191 } else {
192 SignalRecvPacketPreCrypto.disconnect(sink);
193 SignalRecvPacketPreCrypto.connect(sink, OnPacket);
194 }
195 }
196
197 void UnregisterRecvSink(sigslot::has_slots<>* sink,
198 SinkType type) {
199 talk_base::CritScope cs(&signal_recv_packet_cs_);
200 if (SINK_POST_CRYPTO == type) {
201 SignalRecvPacketPostCrypto.disconnect(sink);
202 } else {
203 SignalRecvPacketPreCrypto.disconnect(sink);
204 }
205 }
206
207 bool HasRecvSinks(SinkType type) {
208 talk_base::CritScope cs(&signal_recv_packet_cs_);
209 if (SINK_POST_CRYPTO == type) {
210 return !SignalRecvPacketPostCrypto.is_empty();
211 } else {
212 return !SignalRecvPacketPreCrypto.is_empty();
213 }
214 }
215
216 SsrcMuxFilter* ssrc_filter() { return &ssrc_filter_; }
217
218 const std::vector<StreamParams>& local_streams() const {
219 return local_streams_;
220 }
221 const std::vector<StreamParams>& remote_streams() const {
222 return remote_streams_;
223 }
224
225 // Used for latency measurements.
226 sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
227
228 // Used to alert UI when the muted status changes, perhaps autonomously.
229 sigslot::repeater2<BaseChannel*, bool> SignalAutoMuted;
230
231 // Made public for easier testing.
232 void SetReadyToSend(TransportChannel* channel, bool ready);
233
234 protected:
235 MediaEngineInterface* media_engine() const { return media_engine_; }
236 virtual MediaChannel* media_channel() const { return media_channel_; }
237 void set_rtcp_transport_channel(TransportChannel* transport);
238 bool was_ever_writable() const { return was_ever_writable_; }
239 void set_local_content_direction(MediaContentDirection direction) {
240 local_content_direction_ = direction;
241 }
242 void set_remote_content_direction(MediaContentDirection direction) {
243 remote_content_direction_ = direction;
244 }
245 bool IsReadyToReceive() const;
246 bool IsReadyToSend() const;
247 talk_base::Thread* signaling_thread() { return session_->signaling_thread(); }
248 SrtpFilter* srtp_filter() { return &srtp_filter_; }
249 bool rtcp() const { return rtcp_; }
250
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000251 void FlushRtcpMessages();
252
253 // NetworkInterface implementation, called by MediaEngine
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000254 virtual bool SendPacket(talk_base::Buffer* packet,
255 talk_base::DiffServCodePoint dscp);
256 virtual bool SendRtcp(talk_base::Buffer* packet,
257 talk_base::DiffServCodePoint dscp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000258 virtual int SetOption(SocketType type, talk_base::Socket::Option o, int val);
259
260 // From TransportChannel
261 void OnWritableState(TransportChannel* channel);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000262 virtual void OnChannelRead(TransportChannel* channel,
263 const char* data,
264 size_t len,
265 const talk_base::PacketTime& packet_time,
266 int flags);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000267 void OnReadyToSend(TransportChannel* channel);
268
269 bool PacketIsRtcp(const TransportChannel* channel, const char* data,
270 size_t len);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000271 bool SendPacket(bool rtcp, talk_base::Buffer* packet,
272 talk_base::DiffServCodePoint dscp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000273 virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000274 void HandlePacket(bool rtcp, talk_base::Buffer* packet,
275 const talk_base::PacketTime& packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000276
277 // Apply the new local/remote session description.
278 void OnNewLocalDescription(BaseSession* session, ContentAction action);
279 void OnNewRemoteDescription(BaseSession* session, ContentAction action);
280
281 void EnableMedia_w();
282 void DisableMedia_w();
283 virtual bool MuteStream_w(uint32 ssrc, bool mute);
284 bool IsStreamMuted_w(uint32 ssrc);
285 void ChannelWritable_w();
286 void ChannelNotWritable_w();
287 bool AddRecvStream_w(const StreamParams& sp);
288 bool RemoveRecvStream_w(uint32 ssrc);
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000289 bool AddSendStream_w(const StreamParams& sp);
290 bool RemoveSendStream_w(uint32 ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000291 virtual bool ShouldSetupDtlsSrtp() const;
292 // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
293 // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
294 bool SetupDtlsSrtp(bool rtcp_channel);
295 // Set the DTLS-SRTP cipher policy on this channel as appropriate.
296 bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
297
298 virtual void ChangeState() = 0;
299
300 // Gets the content info appropriate to the channel (audio or video).
301 virtual const ContentInfo* GetFirstContent(
302 const SessionDescription* sdesc) = 0;
303 bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000304 ContentAction action,
305 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000306 bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000307 ContentAction action,
308 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000309 bool SetBaseLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000310 ContentAction action,
311 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000312 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000313 ContentAction action,
314 std::string* error_desc) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000315 bool SetBaseRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000316 ContentAction action,
317 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000318 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000319 ContentAction action,
320 std::string* error_desc) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000321
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000322 // Helper method to get RTP Absoulute SendTime extension header id if
323 // present in remote supported extensions list.
324 void MaybeCacheRtpAbsSendTimeHeaderExtension(
325 const std::vector<RtpHeaderExtension>& extensions);
326
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000327 bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
328 bool* dtls,
329 std::string* error_desc);
330 bool SetSrtp_w(const std::vector<CryptoParams>& params,
331 ContentAction action,
332 ContentSource src,
333 std::string* error_desc);
334 bool SetRtcpMux_w(bool enable,
335 ContentAction action,
336 ContentSource src,
337 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000338
339 // From MessageHandler
340 virtual void OnMessage(talk_base::Message* pmsg);
341
342 // Handled in derived classes
343 // Get the SRTP ciphers to use for RTP media
344 virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const = 0;
345 virtual void OnConnectionMonitorUpdate(SocketMonitor* monitor,
346 const std::vector<ConnectionInfo>& infos) = 0;
347
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000348 // Helper function for invoking bool-returning methods on the worker thread.
349 template <class FunctorT>
350 bool InvokeOnWorker(const FunctorT& functor) {
351 return worker_thread_->Invoke<bool>(functor);
352 }
353
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000354 private:
355 sigslot::signal3<const void*, size_t, bool> SignalSendPacketPreCrypto;
356 sigslot::signal3<const void*, size_t, bool> SignalSendPacketPostCrypto;
357 sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPreCrypto;
358 sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPostCrypto;
359 talk_base::CriticalSection signal_send_packet_cs_;
360 talk_base::CriticalSection signal_recv_packet_cs_;
361
362 talk_base::Thread* worker_thread_;
363 MediaEngineInterface* media_engine_;
364 BaseSession* session_;
365 MediaChannel* media_channel_;
366 std::vector<StreamParams> local_streams_;
367 std::vector<StreamParams> remote_streams_;
368
369 std::string content_name_;
370 bool rtcp_;
371 TransportChannel* transport_channel_;
372 TransportChannel* rtcp_transport_channel_;
373 SrtpFilter srtp_filter_;
374 RtcpMuxFilter rtcp_mux_filter_;
375 SsrcMuxFilter ssrc_filter_;
376 talk_base::scoped_ptr<SocketMonitor> socket_monitor_;
377 bool enabled_;
378 bool writable_;
379 bool rtp_ready_to_send_;
380 bool rtcp_ready_to_send_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000381 bool was_ever_writable_;
382 MediaContentDirection local_content_direction_;
383 MediaContentDirection remote_content_direction_;
384 std::set<uint32> muted_streams_;
385 bool has_received_packet_;
386 bool dtls_keyed_;
387 bool secure_required_;
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000388 int rtp_abs_sendtime_extn_id_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000389};
390
391// VoiceChannel is a specialization that adds support for early media, DTMF,
392// and input/output level monitoring.
393class VoiceChannel : public BaseChannel {
394 public:
395 VoiceChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine,
396 VoiceMediaChannel* channel, BaseSession* session,
397 const std::string& content_name, bool rtcp);
398 ~VoiceChannel();
399 bool Init();
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000400 bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer);
401 bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000402
403 // downcasts a MediaChannel
404 virtual VoiceMediaChannel* media_channel() const {
405 return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
406 }
407
408 bool SetRingbackTone(const void* buf, int len);
409 void SetEarlyMedia(bool enable);
410 // This signal is emitted when we have gone a period of time without
411 // receiving early media. When received, a UI should start playing its
412 // own ringing sound
413 sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
414
415 bool PlayRingbackTone(uint32 ssrc, bool play, bool loop);
416 // TODO(ronghuawu): Replace PressDTMF with InsertDtmf.
417 bool PressDTMF(int digit, bool playout);
418 // Returns if the telephone-event has been negotiated.
419 bool CanInsertDtmf();
420 // Send and/or play a DTMF |event| according to the |flags|.
421 // The DTMF out-of-band signal will be used on sending.
422 // The |ssrc| should be either 0 or a valid send stream ssrc.
henrike@webrtc.org9de257d2013-07-17 14:42:53 +0000423 // The valid value for the |event| are 0 which corresponding to DTMF
424 // event 0-9, *, #, A-D.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000425 bool InsertDtmf(uint32 ssrc, int event_code, int duration, int flags);
426 bool SetOutputScaling(uint32 ssrc, double left, double right);
427 // Get statistics about the current media session.
428 bool GetStats(VoiceMediaInfo* stats);
429
430 // Monitoring functions
431 sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
432 SignalConnectionMonitor;
433
434 void StartMediaMonitor(int cms);
435 void StopMediaMonitor();
436 sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
437
438 void StartAudioMonitor(int cms);
439 void StopAudioMonitor();
440 bool IsAudioMonitorRunning() const;
441 sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;
442
443 void StartTypingMonitor(const TypingMonitorOptions& settings);
444 void StopTypingMonitor();
445 bool IsTypingMonitorRunning() const;
446
447 // Overrides BaseChannel::MuteStream_w.
448 virtual bool MuteStream_w(uint32 ssrc, bool mute);
449
450 int GetInputLevel_w();
451 int GetOutputLevel_w();
452 void GetActiveStreams_w(AudioInfo::StreamList* actives);
453
454 // Signal errors from VoiceMediaChannel. Arguments are:
455 // ssrc(uint32), and error(VoiceMediaChannel::Error).
456 sigslot::signal3<VoiceChannel*, uint32, VoiceMediaChannel::Error>
457 SignalMediaError;
458
459 // Configuration and setting.
460 bool SetChannelOptions(const AudioOptions& options);
461
462 private:
463 // overrides from BaseChannel
464 virtual void OnChannelRead(TransportChannel* channel,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000465 const char* data, size_t len,
466 const talk_base::PacketTime& packet_time,
467 int flags);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000468 virtual void ChangeState();
469 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
470 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000471 ContentAction action,
472 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000473 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000474 ContentAction action,
475 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000476 bool SetRingbackTone_w(const void* buf, int len);
477 bool PlayRingbackTone_w(uint32 ssrc, bool play, bool loop);
478 void HandleEarlyMediaTimeout();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000479 bool InsertDtmf_w(uint32 ssrc, int event, int duration, int flags);
480 bool SetOutputScaling_w(uint32 ssrc, double left, double right);
481 bool GetStats_w(VoiceMediaInfo* stats);
482
483 virtual void OnMessage(talk_base::Message* pmsg);
484 virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const;
485 virtual void OnConnectionMonitorUpdate(
486 SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos);
487 virtual void OnMediaMonitorUpdate(
488 VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
489 void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
490 void OnVoiceChannelError(uint32 ssrc, VoiceMediaChannel::Error error);
491 void SendLastMediaError();
492 void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000493
494 static const int kEarlyMediaTimeout = 1000;
495 bool received_media_;
496 talk_base::scoped_ptr<VoiceMediaMonitor> media_monitor_;
497 talk_base::scoped_ptr<AudioMonitor> audio_monitor_;
498 talk_base::scoped_ptr<TypingMonitor> typing_monitor_;
499};
500
501// VideoChannel is a specialization for video.
502class VideoChannel : public BaseChannel {
503 public:
504 // Make screen capturer virtual so that it can be overriden in testing.
505 // E.g. used to test that window events are triggered correctly.
506 class ScreenCapturerFactory {
507 public:
508 virtual VideoCapturer* CreateScreenCapturer(const ScreencastId& window) = 0;
509 virtual ~ScreenCapturerFactory() {}
510 };
511
512 VideoChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine,
513 VideoMediaChannel* channel, BaseSession* session,
514 const std::string& content_name, bool rtcp,
515 VoiceChannel* voice_channel);
516 ~VideoChannel();
517 bool Init();
518
519 bool SetRenderer(uint32 ssrc, VideoRenderer* renderer);
520 bool ApplyViewRequest(const ViewRequest& request);
521
522 // TODO(pthatcher): Refactor to use a "capture id" instead of an
523 // ssrc here as the "key".
524 VideoCapturer* AddScreencast(uint32 ssrc, const ScreencastId& id);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000525 bool SetCapturer(uint32 ssrc, VideoCapturer* capturer);
526 bool RemoveScreencast(uint32 ssrc);
527 // True if we've added a screencast. Doesn't matter if the capturer
528 // has been started or not.
529 bool IsScreencasting();
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000530 int GetScreencastFps(uint32 ssrc);
531 int GetScreencastMaxPixels(uint32 ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000532 // Get statistics about the current media session.
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000533 bool GetStats(const StatsOptions& options, VideoMediaInfo* stats);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000534
535 sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&>
536 SignalConnectionMonitor;
537
538 void StartMediaMonitor(int cms);
539 void StopMediaMonitor();
540 sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
541 sigslot::signal2<uint32, talk_base::WindowEvent> SignalScreencastWindowEvent;
542
543 bool SendIntraFrame();
544 bool RequestIntraFrame();
545 sigslot::signal3<VideoChannel*, uint32, VideoMediaChannel::Error>
546 SignalMediaError;
547
548 void SetScreenCaptureFactory(
549 ScreenCapturerFactory* screencapture_factory);
550
551 // Configuration and setting.
552 bool SetChannelOptions(const VideoOptions& options);
553
554 protected:
555 // downcasts a MediaChannel
556 virtual VideoMediaChannel* media_channel() const {
557 return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
558 }
559
560 private:
561 typedef std::map<uint32, VideoCapturer*> ScreencastMap;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000562 struct ScreencastDetailsData;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000563
564 // overrides from BaseChannel
565 virtual void ChangeState();
566 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
567 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000568 ContentAction action,
569 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000570 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000571 ContentAction action,
572 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000573 bool ApplyViewRequest_w(const ViewRequest& request);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000574
575 VideoCapturer* AddScreencast_w(uint32 ssrc, const ScreencastId& id);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000576 bool RemoveScreencast_w(uint32 ssrc);
577 void OnScreencastWindowEvent_s(uint32 ssrc, talk_base::WindowEvent we);
578 bool IsScreencasting_w() const;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000579 void GetScreencastDetails_w(ScreencastDetailsData* d) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000580 void SetScreenCaptureFactory_w(
581 ScreenCapturerFactory* screencapture_factory);
582 bool GetStats_w(VideoMediaInfo* stats);
583
584 virtual void OnMessage(talk_base::Message* pmsg);
585 virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const;
586 virtual void OnConnectionMonitorUpdate(
587 SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos);
588 virtual void OnMediaMonitorUpdate(
589 VideoMediaChannel* media_channel, const VideoMediaInfo& info);
590 virtual void OnScreencastWindowEvent(uint32 ssrc,
591 talk_base::WindowEvent event);
592 virtual void OnStateChange(VideoCapturer* capturer, CaptureState ev);
593 bool GetLocalSsrc(const VideoCapturer* capturer, uint32* ssrc);
594
595 void OnVideoChannelError(uint32 ssrc, VideoMediaChannel::Error error);
596 void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000597
598 VoiceChannel* voice_channel_;
599 VideoRenderer* renderer_;
600 talk_base::scoped_ptr<ScreenCapturerFactory> screencapture_factory_;
601 ScreencastMap screencast_capturers_;
602 talk_base::scoped_ptr<VideoMediaMonitor> media_monitor_;
603
604 talk_base::WindowEvent previous_we_;
605};
606
607// DataChannel is a specialization for data.
608class DataChannel : public BaseChannel {
609 public:
610 DataChannel(talk_base::Thread* thread,
611 DataMediaChannel* media_channel,
612 BaseSession* session,
613 const std::string& content_name,
614 bool rtcp);
615 ~DataChannel();
616 bool Init();
617
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000618 virtual bool SendData(const SendDataParams& params,
619 const talk_base::Buffer& payload,
620 SendDataResult* result);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000621
622 void StartMediaMonitor(int cms);
623 void StopMediaMonitor();
624
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000625 // Should be called on the signaling thread only.
626 bool ready_to_send_data() const {
627 return ready_to_send_data_;
628 }
629
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000630 sigslot::signal2<DataChannel*, const DataMediaInfo&> SignalMediaMonitor;
631 sigslot::signal2<DataChannel*, const std::vector<ConnectionInfo>&>
632 SignalConnectionMonitor;
633 sigslot::signal3<DataChannel*, uint32, DataMediaChannel::Error>
634 SignalMediaError;
635 sigslot::signal3<DataChannel*,
636 const ReceiveDataParams&,
637 const talk_base::Buffer&>
638 SignalDataReceived;
639 // Signal for notifying when the channel becomes ready to send data.
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000640 // That occurs when the channel is enabled, the transport is writable,
641 // both local and remote descriptions are set, and the channel is unblocked.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000642 sigslot::signal1<bool> SignalReadyToSendData;
643
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000644 protected:
645 // downcasts a MediaChannel.
646 virtual DataMediaChannel* media_channel() const {
647 return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
648 }
649
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000650 private:
651 struct SendDataMessageData : public talk_base::MessageData {
652 SendDataMessageData(const SendDataParams& params,
653 const talk_base::Buffer* payload,
654 SendDataResult* result)
655 : params(params),
656 payload(payload),
657 result(result),
658 succeeded(false) {
659 }
660
661 const SendDataParams& params;
662 const talk_base::Buffer* payload;
663 SendDataResult* result;
664 bool succeeded;
665 };
666
667 struct DataReceivedMessageData : public talk_base::MessageData {
668 // We copy the data because the data will become invalid after we
669 // handle DataMediaChannel::SignalDataReceived but before we fire
670 // SignalDataReceived.
671 DataReceivedMessageData(
672 const ReceiveDataParams& params, const char* data, size_t len)
673 : params(params),
674 payload(data, len) {
675 }
676 const ReceiveDataParams params;
677 const talk_base::Buffer payload;
678 };
679
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000680 typedef talk_base::TypedMessageData<bool> DataChannelReadyToSendMessageData;
681
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000682 // overrides from BaseChannel
683 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
684 // If data_channel_type_ is DCT_NONE, set it. Otherwise, check that
685 // it's the same as what was set previously. Returns false if it's
686 // set to one type one type and changed to another type later.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000687 bool SetDataChannelType(DataChannelType new_data_channel_type,
688 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000689 // Same as SetDataChannelType, but extracts the type from the
690 // DataContentDescription.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000691 bool SetDataChannelTypeFromContent(const DataContentDescription* content,
692 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000693 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000694 ContentAction action,
695 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000696 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000697 ContentAction action,
698 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000699 virtual void ChangeState();
700 virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet);
701
702 virtual void OnMessage(talk_base::Message* pmsg);
703 virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const;
704 virtual void OnConnectionMonitorUpdate(
705 SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos);
706 virtual void OnMediaMonitorUpdate(
707 DataMediaChannel* media_channel, const DataMediaInfo& info);
708 virtual bool ShouldSetupDtlsSrtp() const;
709 void OnDataReceived(
710 const ReceiveDataParams& params, const char* data, size_t len);
711 void OnDataChannelError(uint32 ssrc, DataMediaChannel::Error error);
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000712 void OnDataChannelReadyToSend(bool writable);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000713 void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
714
715 talk_base::scoped_ptr<DataMediaMonitor> media_monitor_;
716 // TODO(pthatcher): Make a separate SctpDataChannel and
717 // RtpDataChannel instead of using this.
718 DataChannelType data_channel_type_;
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000719 bool ready_to_send_data_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000720};
721
722} // namespace cricket
723
724#endif // TALK_SESSION_MEDIA_CHANNEL_H_