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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>
deadbeefcbecd352015-09-23 11:50:27 -070033#include <map>
34#include <set>
35#include <utility>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037#include "talk/media/base/mediachannel.h"
38#include "talk/media/base/mediaengine.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000039#include "talk/media/base/streamparams.h"
40#include "talk/media/base/videocapturer.h"
deadbeefcbecd352015-09-23 11:50:27 -070041#include "webrtc/p2p/base/transportcontroller.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000042#include "webrtc/p2p/client/socketmonitor.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000043#include "talk/session/media/audiomonitor.h"
buildbot@webrtc.org5ee0f052014-05-05 20:18:08 +000044#include "talk/session/media/bundlefilter.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000045#include "talk/session/media/mediamonitor.h"
46#include "talk/session/media/mediasession.h"
47#include "talk/session/media/rtcpmuxfilter.h"
48#include "talk/session/media/srtpfilter.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000049#include "webrtc/base/asyncudpsocket.h"
50#include "webrtc/base/criticalsection.h"
51#include "webrtc/base/network.h"
52#include "webrtc/base/sigslot.h"
53#include "webrtc/base/window.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000054
55namespace cricket {
56
57struct CryptoParams;
58class MediaContentDescription;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059struct 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
solenberg1dd98f32015-09-10 01:57:14 -070067// enable, marshaling calls to a worker thread, and
henrike@webrtc.org28e20752013-07-10 00:45:36 +000068// 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
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000076 : public rtc::MessageHandler, public sigslot::has_slots<>,
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +000077 public MediaChannel::NetworkInterface,
78 public ConnectionStatsGetter {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000079 public:
deadbeefcbecd352015-09-23 11:50:27 -070080 BaseChannel(rtc::Thread* thread,
81 MediaChannel* channel,
82 TransportController* transport_controller,
83 const std::string& content_name,
84 bool rtcp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085 virtual ~BaseChannel();
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +000086 bool Init();
wu@webrtc.org78187522013-10-07 23:32:02 +000087 // Deinit may be called multiple times and is simply ignored if it's alreay
88 // done.
89 void Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +000090
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000091 rtc::Thread* worker_thread() const { return worker_thread_; }
deadbeefcbecd352015-09-23 11:50:27 -070092 const std::string& content_name() const { return content_name_; }
93 const std::string& transport_name() const { return transport_name_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +000094 TransportChannel* transport_channel() const {
95 return transport_channel_;
96 }
97 TransportChannel* rtcp_transport_channel() const {
98 return rtcp_transport_channel_;
99 }
100 bool enabled() const { return enabled_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000101
102 // This function returns true if we are using SRTP.
103 bool secure() const { return srtp_filter_.IsActive(); }
104 // The following function returns true if we are using
105 // DTLS-based keying. If you turned off SRTP later, however
106 // you could have secure() == false and dtls_secure() == true.
107 bool secure_dtls() const { return dtls_keyed_; }
108 // This function returns true if we require secure channel for call setup.
109 bool secure_required() const { return secure_required_; }
110
111 bool writable() const { return writable_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000112
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700113 // Activate RTCP mux, regardless of the state so far. Once
114 // activated, it can not be deactivated, and if the remote
115 // description doesn't support RTCP mux, setting the remote
116 // description will fail.
117 void ActivateRtcpMux();
deadbeefcbecd352015-09-23 11:50:27 -0700118 bool SetTransport(const std::string& transport_name);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000119 bool PushdownLocalDescription(const SessionDescription* local_desc,
120 ContentAction action,
121 std::string* error_desc);
122 bool PushdownRemoteDescription(const SessionDescription* remote_desc,
123 ContentAction action,
124 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000125 // Channel control
126 bool SetLocalContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000127 ContentAction action,
128 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000129 bool SetRemoteContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000130 ContentAction action,
131 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000132
133 bool Enable(bool enable);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000134
135 // Multiplexing
136 bool AddRecvStream(const StreamParams& sp);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200137 bool RemoveRecvStream(uint32_t ssrc);
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000138 bool AddSendStream(const StreamParams& sp);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200139 bool RemoveSendStream(uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000140
141 // Monitoring
142 void StartConnectionMonitor(int cms);
143 void StopConnectionMonitor();
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000144 // For ConnectionStatsGetter, used by ConnectionMonitor
deadbeefcbecd352015-09-23 11:50:27 -0700145 bool GetConnectionStats(ConnectionInfos* infos) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000146
buildbot@webrtc.org5ee0f052014-05-05 20:18:08 +0000147 BundleFilter* bundle_filter() { return &bundle_filter_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000148
149 const std::vector<StreamParams>& local_streams() const {
150 return local_streams_;
151 }
152 const std::vector<StreamParams>& remote_streams() const {
153 return remote_streams_;
154 }
155
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000156 sigslot::signal2<BaseChannel*, bool> SignalDtlsSetupFailure;
157 void SignalDtlsSetupFailure_w(bool rtcp);
158 void SignalDtlsSetupFailure_s(bool rtcp);
159
buildbot@webrtc.org6bfd6192014-05-15 16:15:59 +0000160 // Used for latency measurements.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000161 sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
162
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000163 // Made public for easier testing.
deadbeefcbecd352015-09-23 11:50:27 -0700164 void SetReadyToSend(bool rtcp, bool ready);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000165
guoweis@webrtc.org4f852882015-03-12 20:09:44 +0000166 // Only public for unit tests. Otherwise, consider protected.
167 virtual int SetOption(SocketType type, rtc::Socket::Option o, int val);
168
solenberg5b14b422015-10-01 04:10:31 -0700169 SrtpFilter* srtp_filter() { return &srtp_filter_; }
170
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000171 protected:
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000172 virtual MediaChannel* media_channel() const { return media_channel_; }
deadbeefcbecd352015-09-23 11:50:27 -0700173 // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
174 // true). Gets the transport channels from |transport_controller_|.
175 bool SetTransport_w(const std::string& transport_name);
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000176 void set_transport_channel(TransportChannel* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000177 void set_rtcp_transport_channel(TransportChannel* transport);
178 bool was_ever_writable() const { return was_ever_writable_; }
179 void set_local_content_direction(MediaContentDirection direction) {
180 local_content_direction_ = direction;
181 }
182 void set_remote_content_direction(MediaContentDirection direction) {
183 remote_content_direction_ = direction;
184 }
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700185 void set_secure_required(bool secure_required) {
186 secure_required_ = secure_required;
187 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000188 bool IsReadyToReceive() const;
189 bool IsReadyToSend() const;
deadbeefcbecd352015-09-23 11:50:27 -0700190 rtc::Thread* signaling_thread() {
191 return transport_controller_->signaling_thread();
192 }
deadbeefcbecd352015-09-23 11:50:27 -0700193 bool rtcp_transport_enabled() const { return rtcp_transport_enabled_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000194
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000195 void ConnectToTransportChannel(TransportChannel* tc);
196 void DisconnectFromTransportChannel(TransportChannel* tc);
197
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000198 void FlushRtcpMessages();
199
200 // NetworkInterface implementation, called by MediaEngine
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000201 virtual bool SendPacket(rtc::Buffer* packet,
stefanc1aeaf02015-10-15 07:26:07 -0700202 const rtc::PacketOptions& options);
203 virtual bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000204
205 // From TransportChannel
206 void OnWritableState(TransportChannel* channel);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000207 virtual void OnChannelRead(TransportChannel* channel,
208 const char* data,
209 size_t len,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000210 const rtc::PacketTime& packet_time,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000211 int flags);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000212 void OnReadyToSend(TransportChannel* channel);
213
214 bool PacketIsRtcp(const TransportChannel* channel, const char* data,
215 size_t len);
stefanc1aeaf02015-10-15 07:26:07 -0700216 bool SendPacket(bool rtcp,
217 rtc::Buffer* packet,
218 const rtc::PacketOptions& options);
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000219 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
220 void HandlePacket(bool rtcp, rtc::Buffer* packet,
221 const rtc::PacketTime& packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000222
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 void EnableMedia_w();
224 void DisableMedia_w();
deadbeefcbecd352015-09-23 11:50:27 -0700225 void UpdateWritableState_w();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000226 void ChannelWritable_w();
227 void ChannelNotWritable_w();
228 bool AddRecvStream_w(const StreamParams& sp);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200229 bool RemoveRecvStream_w(uint32_t ssrc);
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000230 bool AddSendStream_w(const StreamParams& sp);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200231 bool RemoveSendStream_w(uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000232 virtual bool ShouldSetupDtlsSrtp() const;
233 // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
234 // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
235 bool SetupDtlsSrtp(bool rtcp_channel);
236 // Set the DTLS-SRTP cipher policy on this channel as appropriate.
237 bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
238
239 virtual void ChangeState() = 0;
240
241 // Gets the content info appropriate to the channel (audio or video).
242 virtual const ContentInfo* GetFirstContent(
243 const SessionDescription* sdesc) = 0;
244 bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000245 ContentAction action,
246 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247 bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000248 ContentAction action,
249 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000250 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000251 ContentAction action,
252 std::string* error_desc) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000253 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000254 ContentAction action,
255 std::string* error_desc) = 0;
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700256 bool SetRtpTransportParameters_w(const MediaContentDescription* content,
257 ContentAction action,
258 ContentSource src,
259 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000260
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000261 // Helper method to get RTP Absoulute SendTime extension header id if
262 // present in remote supported extensions list.
263 void MaybeCacheRtpAbsSendTimeHeaderExtension(
stefanc1aeaf02015-10-15 07:26:07 -0700264 const std::vector<RtpHeaderExtension>& extensions);
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000265
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000266 bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
267 bool* dtls,
268 std::string* error_desc);
269 bool SetSrtp_w(const std::vector<CryptoParams>& params,
270 ContentAction action,
271 ContentSource src,
272 std::string* error_desc);
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700273 void ActivateRtcpMux_w();
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000274 bool SetRtcpMux_w(bool enable,
275 ContentAction action,
276 ContentSource src,
277 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000278
279 // From MessageHandler
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000280 virtual void OnMessage(rtc::Message* pmsg);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000281
282 // Handled in derived classes
283 // Get the SRTP ciphers to use for RTP media
Guo-wei Shieh456696a2015-09-30 21:48:54 -0700284 virtual void GetSrtpCryptoSuiteNames(
285 std::vector<std::string>* ciphers) const = 0;
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000286 virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000287 const std::vector<ConnectionInfo>& infos) = 0;
288
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000289 // Helper function for invoking bool-returning methods on the worker thread.
290 template <class FunctorT>
291 bool InvokeOnWorker(const FunctorT& functor) {
292 return worker_thread_->Invoke<bool>(functor);
293 }
294
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000295 private:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000296 rtc::Thread* worker_thread_;
deadbeefcbecd352015-09-23 11:50:27 -0700297 TransportController* transport_controller_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000298 MediaChannel* media_channel_;
299 std::vector<StreamParams> local_streams_;
300 std::vector<StreamParams> remote_streams_;
301
pthatcher@webrtc.org990a00c2015-03-13 18:20:33 +0000302 const std::string content_name_;
deadbeefcbecd352015-09-23 11:50:27 -0700303 std::string transport_name_;
304 bool rtcp_transport_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000305 TransportChannel* transport_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700306 std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307 TransportChannel* rtcp_transport_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700308 std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000309 SrtpFilter srtp_filter_;
310 RtcpMuxFilter rtcp_mux_filter_;
buildbot@webrtc.org5ee0f052014-05-05 20:18:08 +0000311 BundleFilter bundle_filter_;
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000312 rtc::scoped_ptr<ConnectionMonitor> connection_monitor_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000313 bool enabled_;
314 bool writable_;
315 bool rtp_ready_to_send_;
316 bool rtcp_ready_to_send_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000317 bool was_ever_writable_;
318 MediaContentDirection local_content_direction_;
319 MediaContentDirection remote_content_direction_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000320 bool has_received_packet_;
321 bool dtls_keyed_;
322 bool secure_required_;
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000323 int rtp_abs_sendtime_extn_id_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000324};
325
326// VoiceChannel is a specialization that adds support for early media, DTMF,
327// and input/output level monitoring.
328class VoiceChannel : public BaseChannel {
329 public:
deadbeefcbecd352015-09-23 11:50:27 -0700330 VoiceChannel(rtc::Thread* thread,
331 MediaEngineInterface* media_engine,
332 VoiceMediaChannel* channel,
333 TransportController* transport_controller,
334 const std::string& content_name,
335 bool rtcp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000336 ~VoiceChannel();
337 bool Init();
solenberg1dd98f32015-09-10 01:57:14 -0700338
339 // Configure sending media on the stream with SSRC |ssrc|
340 // If there is only one sending stream SSRC 0 can be used.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200341 bool SetAudioSend(uint32_t ssrc,
solenbergdfc8f4f2015-10-01 02:31:10 -0700342 bool enable,
deadbeefcbecd352015-09-23 11:50:27 -0700343 const AudioOptions* options,
solenberg1dd98f32015-09-10 01:57:14 -0700344 AudioRenderer* renderer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000345
346 // downcasts a MediaChannel
347 virtual VoiceMediaChannel* media_channel() const {
348 return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
349 }
350
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000351 void SetEarlyMedia(bool enable);
352 // This signal is emitted when we have gone a period of time without
353 // receiving early media. When received, a UI should start playing its
354 // own ringing sound
355 sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
356
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000357 // TODO(ronghuawu): Replace PressDTMF with InsertDtmf.
358 bool PressDTMF(int digit, bool playout);
359 // Returns if the telephone-event has been negotiated.
360 bool CanInsertDtmf();
361 // Send and/or play a DTMF |event| according to the |flags|.
362 // The DTMF out-of-band signal will be used on sending.
363 // The |ssrc| should be either 0 or a valid send stream ssrc.
henrike@webrtc.org9de257d2013-07-17 14:42:53 +0000364 // The valid value for the |event| are 0 which corresponding to DTMF
365 // event 0-9, *, #, A-D.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200366 bool InsertDtmf(uint32_t ssrc, int event_code, int duration, int flags);
solenberg4bac9c52015-10-09 02:32:53 -0700367 bool SetOutputVolume(uint32_t ssrc, double volume);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000368 // Get statistics about the current media session.
369 bool GetStats(VoiceMediaInfo* stats);
370
371 // Monitoring functions
372 sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
373 SignalConnectionMonitor;
374
375 void StartMediaMonitor(int cms);
376 void StopMediaMonitor();
377 sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
378
379 void StartAudioMonitor(int cms);
380 void StopAudioMonitor();
381 bool IsAudioMonitorRunning() const;
382 sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;
383
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000384 int GetInputLevel_w();
385 int GetOutputLevel_w();
386 void GetActiveStreams_w(AudioInfo::StreamList* actives);
387
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000388 private:
389 // overrides from BaseChannel
390 virtual void OnChannelRead(TransportChannel* channel,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000391 const char* data, size_t len,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000392 const rtc::PacketTime& packet_time,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000393 int flags);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000394 virtual void ChangeState();
395 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
396 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000397 ContentAction action,
398 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000399 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000400 ContentAction action,
401 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000402 void HandleEarlyMediaTimeout();
Peter Boström0c4e06b2015-10-07 12:23:21 +0200403 bool InsertDtmf_w(uint32_t ssrc, int event, int duration, int flags);
solenberg4bac9c52015-10-09 02:32:53 -0700404 bool SetOutputVolume_w(uint32_t ssrc, double volume);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000405 bool GetStats_w(VoiceMediaInfo* stats);
406
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000407 virtual void OnMessage(rtc::Message* pmsg);
Guo-wei Shieh456696a2015-09-30 21:48:54 -0700408 virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000409 virtual void OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000410 ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000411 virtual void OnMediaMonitorUpdate(
412 VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
413 void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000414
415 static const int kEarlyMediaTimeout = 1000;
Fredrik Solenberg0c022642015-08-05 12:25:22 +0200416 MediaEngineInterface* media_engine_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000417 bool received_media_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000418 rtc::scoped_ptr<VoiceMediaMonitor> media_monitor_;
419 rtc::scoped_ptr<AudioMonitor> audio_monitor_;
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700420
421 // Last AudioSendParameters sent down to the media_channel() via
422 // SetSendParameters.
423 AudioSendParameters last_send_params_;
424 // Last AudioRecvParameters sent down to the media_channel() via
425 // SetRecvParameters.
426 AudioRecvParameters last_recv_params_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000427};
428
429// VideoChannel is a specialization for video.
430class VideoChannel : public BaseChannel {
431 public:
deadbeefcbecd352015-09-23 11:50:27 -0700432 VideoChannel(rtc::Thread* thread,
433 VideoMediaChannel* channel,
434 TransportController* transport_controller,
435 const std::string& content_name,
Fredrik Solenberg0c022642015-08-05 12:25:22 +0200436 bool rtcp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000437 ~VideoChannel();
438 bool Init();
439
Fredrik Solenberg4b60c732015-05-07 14:07:48 +0200440 // downcasts a MediaChannel
441 virtual VideoMediaChannel* media_channel() const {
442 return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
443 }
444
Peter Boström0c4e06b2015-10-07 12:23:21 +0200445 bool SetRenderer(uint32_t ssrc, VideoRenderer* renderer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000446 bool ApplyViewRequest(const ViewRequest& request);
447
448 // TODO(pthatcher): Refactor to use a "capture id" instead of an
449 // ssrc here as the "key".
buildbot@webrtc.org65b98d12014-08-07 22:09:08 +0000450 // Passes ownership of the capturer to the channel.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200451 bool AddScreencast(uint32_t ssrc, VideoCapturer* capturer);
452 bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer);
453 bool RemoveScreencast(uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000454 // True if we've added a screencast. Doesn't matter if the capturer
455 // has been started or not.
456 bool IsScreencasting();
Peter Boström0c4e06b2015-10-07 12:23:21 +0200457 int GetScreencastFps(uint32_t ssrc);
458 int GetScreencastMaxPixels(uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000459 // Get statistics about the current media session.
pbos@webrtc.org058b1f12015-03-04 08:54:32 +0000460 bool GetStats(VideoMediaInfo* stats);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000461
462 sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&>
463 SignalConnectionMonitor;
464
465 void StartMediaMonitor(int cms);
466 void StopMediaMonitor();
467 sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200468 sigslot::signal2<uint32_t, rtc::WindowEvent> SignalScreencastWindowEvent;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000469
470 bool SendIntraFrame();
471 bool RequestIntraFrame();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000472
Peter Boström0c4e06b2015-10-07 12:23:21 +0200473 bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000474
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000475 private:
Peter Boström0c4e06b2015-10-07 12:23:21 +0200476 typedef std::map<uint32_t, VideoCapturer*> ScreencastMap;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000477 struct ScreencastDetailsData;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000478
479 // overrides from BaseChannel
480 virtual void ChangeState();
481 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
482 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000483 ContentAction action,
484 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000485 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000486 ContentAction action,
487 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000488 bool ApplyViewRequest_w(const ViewRequest& request);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000489
Peter Boström0c4e06b2015-10-07 12:23:21 +0200490 bool AddScreencast_w(uint32_t ssrc, VideoCapturer* capturer);
491 bool RemoveScreencast_w(uint32_t ssrc);
492 void OnScreencastWindowEvent_s(uint32_t ssrc, rtc::WindowEvent we);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000493 bool IsScreencasting_w() const;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000494 void GetScreencastDetails_w(ScreencastDetailsData* d) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000495 bool GetStats_w(VideoMediaInfo* stats);
496
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000497 virtual void OnMessage(rtc::Message* pmsg);
Guo-wei Shieh456696a2015-09-30 21:48:54 -0700498 virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000499 virtual void OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000500 ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000501 virtual void OnMediaMonitorUpdate(
502 VideoMediaChannel* media_channel, const VideoMediaInfo& info);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200503 virtual void OnScreencastWindowEvent(uint32_t ssrc, rtc::WindowEvent event);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000504 virtual void OnStateChange(VideoCapturer* capturer, CaptureState ev);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200505 bool GetLocalSsrc(const VideoCapturer* capturer, uint32_t* ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000506
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000507 VideoRenderer* renderer_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000508 ScreencastMap screencast_capturers_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000509 rtc::scoped_ptr<VideoMediaMonitor> media_monitor_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000510
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000511 rtc::WindowEvent previous_we_;
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700512
513 // Last VideoSendParameters sent down to the media_channel() via
514 // SetSendParameters.
515 VideoSendParameters last_send_params_;
516 // Last VideoRecvParameters sent down to the media_channel() via
517 // SetRecvParameters.
518 VideoRecvParameters last_recv_params_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000519};
520
521// DataChannel is a specialization for data.
522class DataChannel : public BaseChannel {
523 public:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000524 DataChannel(rtc::Thread* thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000525 DataMediaChannel* media_channel,
deadbeefcbecd352015-09-23 11:50:27 -0700526 TransportController* transport_controller,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000527 const std::string& content_name,
528 bool rtcp);
529 ~DataChannel();
530 bool Init();
531
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000532 virtual bool SendData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000533 const rtc::Buffer& payload,
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000534 SendDataResult* result);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000535
536 void StartMediaMonitor(int cms);
537 void StopMediaMonitor();
538
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000539 // Should be called on the signaling thread only.
540 bool ready_to_send_data() const {
541 return ready_to_send_data_;
542 }
543
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000544 sigslot::signal2<DataChannel*, const DataMediaInfo&> SignalMediaMonitor;
545 sigslot::signal2<DataChannel*, const std::vector<ConnectionInfo>&>
546 SignalConnectionMonitor;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200547 sigslot::signal3<DataChannel*, const ReceiveDataParams&, const rtc::Buffer&>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000548 SignalDataReceived;
549 // Signal for notifying when the channel becomes ready to send data.
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000550 // That occurs when the channel is enabled, the transport is writable,
551 // both local and remote descriptions are set, and the channel is unblocked.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000552 sigslot::signal1<bool> SignalReadyToSendData;
buildbot@webrtc.org1d66be22014-05-29 22:54:24 +0000553 // Signal for notifying that the remote side has closed the DataChannel.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200554 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000555
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000556 protected:
557 // downcasts a MediaChannel.
558 virtual DataMediaChannel* media_channel() const {
559 return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
560 }
561
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000562 private:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000563 struct SendDataMessageData : public rtc::MessageData {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000564 SendDataMessageData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000565 const rtc::Buffer* payload,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000566 SendDataResult* result)
567 : params(params),
568 payload(payload),
569 result(result),
570 succeeded(false) {
571 }
572
573 const SendDataParams& params;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000574 const rtc::Buffer* payload;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000575 SendDataResult* result;
576 bool succeeded;
577 };
578
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000579 struct DataReceivedMessageData : public rtc::MessageData {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000580 // We copy the data because the data will become invalid after we
581 // handle DataMediaChannel::SignalDataReceived but before we fire
582 // SignalDataReceived.
583 DataReceivedMessageData(
584 const ReceiveDataParams& params, const char* data, size_t len)
585 : params(params),
586 payload(data, len) {
587 }
588 const ReceiveDataParams params;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000589 const rtc::Buffer payload;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000590 };
591
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000592 typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData;
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000593
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000594 // overrides from BaseChannel
595 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
596 // If data_channel_type_ is DCT_NONE, set it. Otherwise, check that
597 // it's the same as what was set previously. Returns false if it's
598 // set to one type one type and changed to another type later.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000599 bool SetDataChannelType(DataChannelType new_data_channel_type,
600 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000601 // Same as SetDataChannelType, but extracts the type from the
602 // DataContentDescription.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000603 bool SetDataChannelTypeFromContent(const DataContentDescription* content,
604 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000605 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000606 ContentAction action,
607 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000608 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000609 ContentAction action,
610 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000611 virtual void ChangeState();
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000612 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000613
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000614 virtual void OnMessage(rtc::Message* pmsg);
Guo-wei Shieh456696a2015-09-30 21:48:54 -0700615 virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000616 virtual void OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000617 ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000618 virtual void OnMediaMonitorUpdate(
619 DataMediaChannel* media_channel, const DataMediaInfo& info);
620 virtual bool ShouldSetupDtlsSrtp() const;
621 void OnDataReceived(
622 const ReceiveDataParams& params, const char* data, size_t len);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200623 void OnDataChannelError(uint32_t ssrc, DataMediaChannel::Error error);
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000624 void OnDataChannelReadyToSend(bool writable);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200625 void OnStreamClosedRemotely(uint32_t sid);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000626
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000627 rtc::scoped_ptr<DataMediaMonitor> media_monitor_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000628 // TODO(pthatcher): Make a separate SctpDataChannel and
629 // RtpDataChannel instead of using this.
630 DataChannelType data_channel_type_;
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000631 bool ready_to_send_data_;
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700632
633 // Last DataSendParameters sent down to the media_channel() via
634 // SetSendParameters.
635 DataSendParameters last_send_params_;
636 // Last DataRecvParameters sent down to the media_channel() via
637 // SetRecvParameters.
638 DataRecvParameters last_recv_params_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000639};
640
641} // namespace cricket
642
643#endif // TALK_SESSION_MEDIA_CHANNEL_H_